Servetas, Andrew
2006-Sep-06 13:35 UTC
[asterisk-users] Volume events causing talk off on Asterisk with Digium 411P
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. 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. Has anyone else experienced this, and if so, what have you done to correct it? Andy Servetas CTI Support Engineer Dirigosoft Corporation Portland, ME www.dirigosoft.com <http://www.dirigosoft.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060906/09150c97/attachment.htm
Zoa
2006-Sep-07 00:36 UTC
[asterisk-users] Volume events causing talk off on Asterisk with Digium 411P
I have the same problem on on of our systems, but i always thought it to be a problem in the ATA's connected to this server. (My customer has a lot of traffic on the lines and only sometimes hears this problem). It seemed to happen especially with loud woman voices, but i was unable to reproduce it on command. I have several other te410p's on different locations (with different carriers), without those complaints. Does this also happen on pri to pri calls for you ? Maybe its a combination of carrier volume with the te410p ? Zoa Servetas, Andrew wrote:> > > 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. > > > > 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. > > > > Has anyone else experienced this, and if so, what have you done to > correct it? > > > > //Andy Servetas// > > CTI Support Engineer > > > > Dirigosoft Corporation > > Portland, ME > > > > www.dirigosoft.com <http://www.dirigosoft.com/> > > > >------------------------------------------------------------------------ > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Servetas, Andrew
2006-Sep-07 06:00 UTC
[asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P
Yes, it seems to be happening on any call that passes over the T1 card. SIP-to-SIP works fine. Date: Thu, 07 Sep 2006 10:36:24 +0300 From: Zoa <zoachien@securax.org> Subject: Re: [asterisk-users] Volume events causing talk off on Asterisk with Digium 411P To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <44FFCBF8.2020908@securax.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed I have the same problem on on of our systems, but i always thought it to be a problem in the ATA's connected to this server. (My customer has a lot of traffic on the lines and only sometimes hears this problem). It seemed to happen especially with loud woman voices, but i was unable to reproduce it on command. I have several other te410p's on different locations (with different carriers), without those complaints. Does this also happen on pri to pri calls for you ? Maybe its a combination of carrier volume with the te410p ? Zoa Servetas, Andrew wrote:> > > 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. > > > > 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. > > > > Has anyone else experienced this, and if so, what have you done to > correct it? > > > > //Andy Servetas// > > CTI Support Engineer > > > > Dirigosoft Corporation > > Portland, ME > > > > www.dirigosoft.com <http://www.dirigosoft.com/> > > >
Steven
2006-Sep-07 06:43 UTC
[asterisk-users] Re: Volume events causing talk off on Asterisk withDigium 411P
What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ?
--
--
Steven
http://www.glimasoutheast.org
"Servetas, Andrew" <andrew.servetas@dirigosoft.com> wrote in
message news:28289145AD231E418DB8CABE0BE392AA016F8176@casco.stroudwater.net...
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.
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.
Has anyone else experienced this, and if so, what have you done to correct it?
Andy Servetas
CTI Support Engineer
Dirigosoft Corporation
Portland, ME
www.dirigosoft.com
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Servetas, Andrew
2006-Sep-07 11:20 UTC
[asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P
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.
_____
What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ?
--
--
Steven
http://www.glimasoutheast.org
"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>
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.
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.
Has anyone else experienced this, and if so, what have you done to
correct it?
Andy Servetas
CTI Support Engineer
Dirigosoft Corporation
Portland, ME
www.dirigosoft.com
------------------------------------------------------------------------
------
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Zoa
2006-Sep-07 11:48 UTC
[asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P
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. > > ------------------------------------------------------------------------ > >What were the proposed changes to VPM_DEFAULT_DTMFTHRESHOLD ? > > > >-- > >-- > >Steven > > > >http://www.glimasoutheast.org > > > > > > > > "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> > > > > > > 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. > > > > > > > > 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. > > > > > > > > Has anyone else experienced this, and if so, what have you done to correct it? > > > > > > > > Andy Servetas > > > > CTI Support Engineer > > > > > > > > Dirigosoft Corporation > > > > Portland, ME > > > > > > > > www.dirigosoft.com > > > > > > > > > > > >------------------------------------------------------------------------------ > > > >------------------------------------------------------------------------ > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Servetas, Andrew
2006-Sep-07 14:47 UTC
[asterisk-users] RE: Volume events causing talk off on Asterisk with Digium 411P
It did help some. I first went to a value of 500, that had no effect. Then I went to 2000 and we had no events for at least 3 days. I then went to 3000 and it got worse again. So now I'm back to 2000, and we're seemingly stable, but not completely rid of the events. I will set up a monitor app to record voice traffic and find some examples of it happening. I will need to trim the audio clips, as we wont want to have full conversations available because of privacy reasons. I will fond something ambiguous and appropriate for distribution so we can try to debug this. Also, I went into logger.conf and enabled DTMF in the console log. When this happens, I DO see a DTMF event in the middle of conversations when no keys are pressed. I will post examples of them here: Sep 7 17:11:51 DTMF[26078]: channel.c:2298 ast_write: SIP/213-67f7 : A Sep 7 17:13:25 DTMF[26078]: channel.c:2298 ast_write: SIP/213-67f7 : 1 Sep 7 17:15:56 DTMF[26078]: channel.c:2298 ast_write: SIP/213-67f7 : 5 As you can see by the timestamps, they happen randomly throughout some calls. ------------------------------ Message: 7 Date: Thu, 07 Sep 2006 21:48:06 +0300 From: Zoa <zoachien@securax.org> Subject: Re: [asterisk-users] Re: Volume events causing talk off on Asterisk with Digium 411P To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <45006966.3090409@securax.org> Content-Type: text/plain; charset=windows-1252; format=flowed 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. >