There is no tx/rxgain on a sip call (other then on the sip phones). Also, no echocancel on sip-->sip calls (unless you turn on when bridged)... but I believe he has stated he is already doing this.
On 9/29/05, Matt <[EMAIL PROTECTED]> wrote: > hi: > > We are using 1.0.9 * with sangoma 104 quad card, hooked to 4 E1s. We have no > echo problems at all. > > The voice qualities sound and clear, try adjust tx/rxgain a bit. and make > sure your zapata.conf's echocancel param is enabled. > > Best Regards > > Matt > ____________________________________________ > High Performance Gigabit Clustering Appliance > http://www.xgforce.com/loadbalancer.html > > eClustering Service > http://www.xgforce.com/eService.html > > Gigabit 3U Tera Servers > http://www.xgforce.com/teraserver.html > > Gigabit 2U Servers > http://www.xgforce.com/server2u.html > > Gigabit 1U Servers > http://www.xgforce.com/server1u.html > > __________________________________________________ > > ----- Original Message ----- > From: "Tom Hayden" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Thursday, September 29, 2005 6:02 AM > Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help, > > > > What kind of POTS trunks/cards are you using? > > > > -- > > Tom > > > > On 9/29/05, Ian Bonham <[EMAIL PROTECTED]> wrote: > > > Hi all, > > > > > > I hope someone can help, as I have an urgent problem. > > > > > > I've got a production Asterisk server thats been deployed, but we are > seeing > > > a strange voice echo problem. There is about a 250ms echo for the users > in > > > the office, and they are hearing their own voice back at them. > > > > > > I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of > > > memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel > > > 2000w handsets, and X-Lite (free) PC clients. All see the same problem. > > > There is a bridge into the POTS (BT's SystemX) using a Voicetronix > > > OpenSwitch12 card and the vpbhp driver. > > > > > > The echo occurs on both SIP->POTS calls, and SIP->SIP calls. I've tried > a > > > number of volume adjustments to correct the echo but it is always the > same. > > > > > > If anyone has any ideas I'd really appriciate some help, as this is a > major > > > urgency, > > > > > > Many many thanks, > > > > > > Ian Bonham > > > > > > _________________________________________________________________ > > > FREE pop-up blocking with the new MSN Toolbar - get it now! > > > http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ > > > > > > _______________________________________________ > > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > [email protected] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > Tom > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
