I use Asterisk 1.4 + FreePBX on Ubuntu x64 10.04.3.
My question is: How can I set Asterisk/FreePBX, so it passes through SDP and RTP? How to disable any codec transcoding?
Now I have some SIP extensions and some SIP trunks defined in FreePBX. Everything works ok, I can make calls.
But now it works this way: if “extension-Asterisk” leg has different codec-preference than “Asterisk-trunk” leg, Asterisk transcodes the audio. It causes big CPU load.
I would like to disable such functionality, and force Asterisk to simply pass-through codec order and description (SDP) from extension to trunk. This way Asterisk would simply pass the RTP from extension to trunk, without transcoding it.
How to achieve it?