ConfBridge Audio Quality

I started to use ConfBridge in Asterisk 11 and noticed that the sound is getting clipped.
Get real static at the top of the volume curve. I connect to the Conference via a sip client running g.722 and I almost have to turn my microphone all the way down, or it will clip. The callers come in on a g.729 and their is a slight clip at the top end when they get loud.

Is there a way to set default levels? on my sip callers vs my DID callers?
Any ideas or help would be great.


There’s a dialplan volume function: … ion_VOLUME

There’s also an automatic gain control function (that utilizes Speex): … nction_AGC