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
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