[Asterisk-Users] sip + zap problem

[joe at jsci.net]

Here's our config:

cisco 7960's running 6.3 sip code
latest cvs of *
t100p zaptel card
adit 600 channel bank
7 pots lines and 2 fax machines on the adit 600

dialing out from the cisco phones gets sent out via the zap channels, but
I'm having some serious echo problems.  I currently have the adit set to
+3 rxgain and -6 txgain, with my zapata.conf containing:

echocancel=128
echocancelwhenbridged=no
rxgain=9.0
txgain=-4.0
jitterbuffers=15
echotraining=no

on the appropriate pots channels.  Now, the received audio is still a bit
low, and the audio I'm sending out is still a little high.  I've tried 32,
64, 128, and 256 on the echocancel, yes and no for when bridged, and an
endless list of different settings on the gains.  I've also tried the echo
training, and all 5 different echo cancelers, even the agressive option in
mark2.  Some configurations had better results than others, but right now
its the best it's been, but I still get a tiny after-sound, sounding kind
of like a robot, on certain sounds and volumes of noise, as if it were an
echo that wasn't fully canceled...

Is anyone else running this kind of config?  If so, do you have/did you
have this kind of problem?  and what did you do to make it work?

My customer needs everything to be up and running correctly by next week,
and I fear I may wind up swapping out his ip phones with analog phones... 
I am willing to pay anyone who can help me get this resolved.

Thanks.

-Joe