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 <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Volume events causing talk off on
Asterisk with Digium 411P
To: Asterisk Users Mailing List - Non-Commercial Discussion
<[email protected]>
Message-ID: <[EMAIL PROTECTED]>
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/>
>
>
>
_______________________________________________
--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
