Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo.
I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN through the FXO cards I get horrible echo, I have even been able when talking loud enough to get a horrible feedback loop going. I have tried 4 different echo cancellers in the Makefile for the Zap drivers and nonoe of them changed the situation. I have echocancel = (Any where from 1 - 256, I have tried alot of different values), and I have echocanelwhenbridged = yes.I only hear the echo start when the call gets bridged onto the outgoing PSTN lines. Is there anything I can do? Brian J. Schrock _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
