Hello everyone,<br>I am new to asterisk and I was wondering - are there any methods of manipulating audio inside asterisk?<br>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?<br>
<br>Kind Regards,<br>Sławomir Słoma<br>