I can minimize doing those tricks, but I cannot seem to get it to go away. On Thu, 2003-08-28 at 11:33, Dan wrote: > ----- 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
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