[asterisk-users] Bad audio quality on SIP

[Asterisk]

Hi guys,

I have the following configuration:

10 SIP softphones <--> Asterisk <--> PSTN

Audio is always good on SIP softphone side, but callers from PSTN side
*sometimes* complain that the audio quality is bad (and volume low). The
QoS is turned on on the computers where SIP softphone is installed, and
the tos setting in sip.conf is set to 0x18.

The interesting thing is that usually only one SIP softphone user (but
each time someone else) would get complaints like that ... others seem
to work okay.

What could be wrong?

Thanx,
Alex