Ever since updating through cvs I have also noticed echo on the pstn lines using a fxo channel bank. I had no echo prior to the cvs update. I have backed off the gain for the FXO channels also (from 2.0 to 0.0) and this helped but there is still sometimes (but not all the time) a very distinct echo. It is like sometimes the echo canceling works and sometimes it does not.
I am now running asterisk cvs-08/26/03-14:04:16. I also did a cvs update for Zaptel and libpri on August 26th since * would not compile without it. Previously I was running the cvs version that I downloaded on April 16, 2003. I am not using a sip phone but an ADSI analog set so it appears that this is not sip related. If someone could verify this, a bug could be filed in bugtracker. Don Pobanz On Thursday, August 28, 2003 10:34 AM, Dan [SMTP:[EMAIL PROTECTED] 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
