I ran that this morning. I'll let it run through today. If it's not good enough then I'll try ASLEC.
Thanks guys On Sat, Oct 10, 2009 at 10:47 AM, Matt Sales <msa...@gmail.com> 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)<http://en.wikipedia.org/wiki/Echo_%28phenomenon%29> >>> >>> 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 >> > > -- Bob Crandell Assured Computing, Inc. 541-868-0331 ComputerBase 541-349-0404
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