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? _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
