[asterisk-dev] Question about interface to sound processing library

[Rolf-Werner Eilert]

Hi folks,

I hope I'm right here in this list. Tried to ask this as a general 
question in forum General first, but there was only a vague answer. Here 
I expect to find the guys who make the core functions of Asterisk, so I 
ask my question again.

To describe the reason for my question: We are running a school for 
foreign languages training students for office communication. This 
includes telephoning in foreign languages. Up to now, we provide a 
simple one-box unit with a wireless phone for the person that leaves the 
classroom and a loudspeaker for the class to listen.

There was the idea of building a telephone system that allows to 
simulate telephone calls to far destinations and to cellphones offering 
kind of simulating distortions typical to such calls (delays, cracks, 
echos, or scenarios like "cellphone at a busy street café" :D ).

With Asterisk, my phantasy goes to offer the trainer an easy interface 
(which I could program myself) to choose line quality, scenarios etc. 
and to have a telephone in every room to call from and to be called. One 
might even simulate international numbers...

My idea is to use some interface to the voice processing modules of 
Asterisk to be able to let them remodulate the sound stream (e. g. 
decreasing the sampling rate for 200 ms or so, then going back to normal 
to simulate typical GSM distortions like organ-like noises) or to mix in 
pre-recorded noises like from a street etc.

Has anyone here ever seen something like this with Asterisk, or are 
there any plugins/modules you would consider worth taking a look at? I 
am new to Asterisk, so I don't know what to look for.

Let me add this: There is a Linux server running 24/7 and a terminal for 
the teachers in every classroom.

Thanks for reading up to here - and thanks a lot for all your ideas!

Rolf