[asterisk-users] Lucent Max TNT PRI Agg --> * --> SIP DEV (PHONE or ATA)

[JR Richardson]

> Hello...
> We're attempting to track down an intermittent echo issue.  Our setup is
> <phone>sip<asterisk>sip<tnt>pri to carriers.  We have less than 2 ms latency on the networks (FTTx), totally SIP w/ G711u.  The party hearing the echo is the subscriber using sip.  The PSTN users does not hear the echo.
>
> We should be note that there is zero echo when calling sip to sip with or without reinvites enabled.
>
> We have several different phones; linksys, polycom, & grandstream (both atas and phones).  It's difficult to reproduce the problem regularly so isolation is an issue.
>

I had intermittent echo when I first deployed TNT's as well.  It took
a while to track down.  Adjust the volume on the TNT lower until the
echo goes away.  Here is what I had to set mine to:

In each T1 config:

set line-interface voip-gain-control input-pad = 3db-loss
set line-interface voip-gain-control output-pad = 3db-loss

Hope this helps.

JR
-- 
JR Richardson
Engineering for the Masses