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

[Steve Underwood]

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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>------- End of Original Message -------
>
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