[Asterisk-Users] SIP echo?

[Richard Alexander]

I have had similar problems but one thing that seemed to help in my case
was to back off the rxgain and txgain for the X100P. I haven't yet had
the chance to experiment fully.

I think I also have echocancel=128 for the X100P channel and the echo
canceller does seem to train up over the first few seconds of a call
now.



-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Joe
Antkowiak
Sent: Friday, May 30, 2003 2:10 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] SIP echo?

I noticed a few other messages posted about this problem, but I couldn't
find an answer...

I'm having a problem with SIP echo when calls are received into asterisk
via an x100p and bridged with a sip extension (back to the pstn with
iconnecthere).  the person calling in to asterisk has no echo problems,
but the recipient of the pstn call, everything they say, they hear back
about 1 second later.  echocancel=yes and echocancelwhenbridged=yes are
in the applicable channels in zapata.conf.  I can also use the PC client
from iconnecthere and I do not have the problem.

Any ideas?

Also, what would be the best codec to use to send fax transmissions via
SIP?
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