[asterisk-users] g.722 + loudness

[Hose]

Hi,                                                                                                                                                                   
                                                                                                                                                                      
We've been running g.722 in asterisk 1.6.09 for awhile now, with a PRI                                                                                                
and numerous sip clients.  Internal sip to sip and sip to pri (and
vice versa) work fine between g.722 and ulaw - the transcoding is                                                                   
acceptable.                                      
                                                           
The only time it fails is when we utilize a meetme conference bridge.
With a Polycom IP 6000 + a call over the PRI, the person calling in over                  
the PRI sounds distorted when they're barely talking at a normal volume.       
Anything over a normal volume results in terrible clipping.  Bringing                                                                                                 
the volume down on the Polycom either via software settings or the                                                                                                    
actual volume keys doesn't stop the distortion, so that points to a                                                                                                   
problem with asterisk (the volume can be very loud, barely audible, but
you can still hear the clipping occuring).  By clipping, I mean the
static that happens when you have a signal that's too loud.

The thing is, when you call directly into the Polycom over the PRI, it's
fine.  This ONLY happens during a conference call with g.722, though
this might be because asterisk is negotiating a ulaw connection when
called direct from the PRI - is there a way to check what codec it's
negotiated during the call?

I have a feeling that the issue is between transcoding of ulaw to g.722
and it's too loud during the transcoding - anyway to adjust the levels?

Thanks!
hose