[Asterisk-Dev] HELP: DTMF signals screwed up by asterisk

[Asterisk Developer]

---------- Original Message -----------
From: Steve Underwood <steveu at coppice.org>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Sent: Fri, 17 Sep 2004 00:21:19 +0800
Subject: Re: [Asterisk-Dev] HELP: DTMF signals screwed up by asterisk

> The 2100Hz tone looks OK, if a bit loud. The level of the DTMF is 
> far too high. It is overloaded, and rather distorted. Something 
> seems wrong with the levels. Do you have the gain of the E1 channel 
> set to something other than the default?
> 
> What is missing is exactly 40ms long. The audio blocks which * 
> processes are 20ms each, so that duration would fit with exactly 2 
> blocks being set to silence. However, that doesn't sound like *'s behaviour.
> 
> Regards,
> Steve
> 
> Asterisk Developer wrote:
> 
> >Steven,
> >
> >thanks. actually that is not true, as it happens with MGCP as well. I'm 
not 
> >using any high compression codec, just plain g711/aLaw. I've recorded the 
> >wave form within asterisk, so the SIP/MGCP leg can't be the problem, it is 
> >definitely the E1/zaptel leg giving me the headache. i've tried to 
generate a 
> >call automatically using the spool/asterisk/outgoing queue, the waveform 
> >still gets deformed/screwed up.
> >
> >bye
> >alf
> >
> >
> >
> >---------- Original Message -----------
> >From: Steven Critchfield <critch at basesys.com>
> >To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> >Sent: Wed, 15 Sep 2004 08:58:35 -0500
> >Subject: Re: [Asterisk-Dev] HELP: DTMF signals screwed up by asterisk
> >
> >  
> >
> >>On Wed, 2004-09-15 at 08:22, Asterisk Developer wrote:
> >>    
> >>
> >>>hello,
> >>>
> >>>I tried to use an analog emergency-call device. that device is a one-
> >>>      
> >>>
> >button 
> >  
> >
> >>>device that calls an emergency service like 911, it's for older people 
or 
> >>>people with disabilities. however, the device dials out to a remote 
> >>>      
> >>>
> >server to 
> >  
> >
> >>>login and gather some information. when the device dials the number, the 
> >>>server answers with a 2000hz tone and a DTMF sequence. that DTMF 
sequence 
> >>>      
> >>>
> >is 
> >  
> >
> >>>however totally screwed up by asterisk. each dtmf sequence is 80ms long, 
> >>>      
> >>>
> >but 
> >  
> >
> >>>asterisk cuts out 40ms from the sequence. it looks like:
> >>>
> >>> *******00000000000000*******
> >>> 0-----20-----40-----60----80---->
> >>> ms                           
> >>>
> >>>* = DTMF signal 
> >>>0 = no DTMF signal
> >>>
> >>>call direction SIP<->ZAP<->E1 (TE410P). I have a WAV file and a PNG 
image 
> >>>      
> >>>
> >of 
> >  
> >
> >>>the waveform. see http://www.jass.at/asterisk/
> >>>
> >>>do we have a chance to turn off DTMF detection within zaptel/chan_zap?
> >>>      
> >>>
> >>You can't pass DTMF inband on the SIP leg if you use any compression.
> >>This means the problem you are experiencing is really in your SIP 
> >>leg of the call. Check your SIP device to see if it will let you 
> >>change the length of the DTMF tones.
> >>-- 
> >>Steven Critchfield <critch at basesys.com>
> >>
> >>_______________________________________________
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> >>    
> >>
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> >
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> >  
> >
> 
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