Re: [asterisk-gui] Echo Canceling

[Greg Nutt]

I personally ended up removing the hardware canceler from our TDM404E and just used OSLEC.  I tried unsuccessfully for a month to get the hardware canceler to work, but no matter what I still ended up with echo is some scenarios.  OSLEC has been the best solution from my experience. Greg Matt Sales wrote: It has been my experience in the past that running fxotune on your system will clear up most echo issues.  Read more about it here: http://www.voip-info.org/wiki/view/Asterisk+fxotune On Fri, Oct 9, 2009 at 3:58 PM, Bob Crandell <rob...@assuredcomp.com> wrote: Andrew, These are POTS lines. If it's glare then what do I change? I'm kind of hoping I can make these adjustments in the GUI.  Is it possible? How do you have yours set? Thanks On Fri, Oct 9, 2009 at 10:55 AM, Andrew Latham <andrew.lat...@tuxtone.com> wrote: Bob On to the fun topic of audio quality....  The (echo, glare, volume) canceler is getting "trained" during your call so that is why it fixes its self.  If you are using POTS for your trunk lines then you will want to work on the DAHDI/Zaptel settings to resolve any issues.  You may be dealing with glare instead of echo. http://en.wikipedia.org/wiki/Echo_(phenomenon) To fix you can enable and fine tune settings in your DAHDI/Zaptel settings files and restart both Asterisk and DAHDI/Zaptel. ~ Andrew "lathama" Latham andrew.lat...@tuxtone.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Fri, Oct 9, 2009 at 1:43 PM, Bob Crandell <rob...@assuredcomp.com> wrote: > Hi, > > My next challenge is canceling the echo when I make or receive a call. > Right now the echo lasts for the first few seconds after the call is > answered then stops. > There is no echo when calling an extension. > Under Configure hardware is software echo canceller.  I tried a couple of > things there that didn't make a noticable difference. > Under Advanced Options/SIP Settings there is Jitter Buffer which might have > made a tiny difference. > > What are you guys doing about this? > > Thanks > -- > Bob Crandell > Assured Computing, Inc. > 541-868-0331 > ComputerBase > 541-349-0404 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: >   http://lists.digium.com/mailman/listinfo/asterisk-gui > -- Bob Crandell Assured Computing, Inc. 541-868-0331 ComputerBase 541-349-0404 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-gui _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui

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