----- Original Message ----- From: "Brian J. Schrock" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 28, 2003 6:16 PM Subject: [Asterisk-Users] SIP and ECHO
> 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 > Hi, For me: rxgain=0.8 txgain=0.8 in zapata conf do the trick. Now the echo is allmost inexistant. Maybe the sound is not very strong but the quality is very good. I have the default echo canceller (no modification in the source files). Tested with a lot of SIP phones (ATA (G.711), X-Lite(GSM), SJ_phone(G.711), Cisco 79x0) and one X100P card. BR, Dan _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
