[asterisk-dev] Manipulating audio in asterisk

[Olle E. Johansson]

21 jan 2010 kl. 09.26 skrev Slawek Sloma:

> Hello everyone,
> I am new to asterisk and I was wondering - are there any methods of manipulating audio inside asterisk?
> What I would like to achive is to manipulate audio waveform inside asterisk. I have seen the func_volume.c file and the audiohooks but it doesn't seem to work with sip protocol - or maybe I am doing something wrong?

Almost everything in the asterisk dialplan works with all protocols, that's the main idea behind Asterisk.

I would encourage you to look at app_jack and the Jack toolkit that it uses. As far as I understand, you can manipulate audio any way you want with it.

For general questions on Asterisk - how to get this working in SIP and such, please use the asterisk-users mailing list.

Regards,
/Olle