[asterisk-users] Re: Volume events causing talk off on Asterisk
with Digium 411P
Zoa
zoachien at securax.org
Thu Sep 7 11:48:06 MST 2006
But does it help ? Is it better than before ?
Do you have a good way of debugging ? (like an audio recording that i
could play ?)
Does it show something on the cli when it happens ?
Zoa
Servetas, Andrew wrote:
> They recommended changing the default value of 1000 up or down
> incrementally until it works better. We’re currently at 2000, and
> we’re still not completely free of events.
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>What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ?
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>Steven
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>http://www.glimasoutheast.org
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> "Servetas, Andrew" <andrew.servetas at dirigosoft.com <http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote in message news:28289145AD231E418DB8CABE0BE392AA016F8176 at casco.stroudwater.net... <http://lists.digium.com/mailman/listinfo/asterisk-users>
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> We are experiencing random talk off events when we hear a loud volume event on the PSTN side of our calls. We do not always hear the spurious DTMF, but I can see it in the console when I have the debug and verbose levels turned up. We do however always have the associated brief periods of silence that immediately follow. Sometimes they are only a matter of seconds, other times they can be as long as a minute. We hear it most often if the remote party is on a cellular phone with a lot of background noise, or if a loud noise happens during the call. Neither party can hear the other when this happens. It almost reacts like an AGC circuit is muting the call.
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> We are using a Digium TE411P quad-span T1 card on 1.2.5. I called Digium support and we have played with the VPM_DEFAULT_DTMFTHRESHOLD in the WCT4XXP.C driver as recommended, and all the RELAXDTMF and GAIN settings in Zapata.conf are set according to their recommendations.
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> Has anyone else experienced this, and if so, what have you done to correct it?
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> Andy Servetas
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> CTI Support Engineer
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> Dirigosoft Corporation
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> Portland, ME
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> www.dirigosoft.com
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