Did you enable echocancel and echocancelwhilebridged? Did you put them in the correct location in the zapata.conf ? It has to be before the channel statement (this is what threw me for a week) If you tail -f debug in the /var/log/asterisk you can watch the call and see if echo cancel was kicking in
Lee ----- Original Message ----- From: "John Brown" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, August 20, 2003 2:23 AM Subject: [Asterisk-Users] echo on the sip side > so i call from a sip phone (grandstream) to > a cell via x100p > > > PSTN side hears everything nice, no echo. > > on the SIP side I hear myself about .1 to .2 sec > later... > > any thoughts on how to resolve this. > > mucho thanks to everyone that has been helpful :) > > john > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
