[Asterisk-Users] sip + zap problem

[joe at jsci.net]

Just had another thought, about replacing the cb and zaptel card with a
sip<>analog gateway...  Can anyone recommend one?  (in case I can't get
this straightened out)

> 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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>