[asterisk-users] Double DTMF digits

[Dovid B]

I am actually getting DTMF over SIP when people call in to a clients system 
that is running a2billing. They are using RFC2833.

----- Original Message ----- 
From: "Remi Quezada" <remiq at monmouth.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Wednesday, May 09, 2007 6:15 PM
Subject: Re: [asterisk-users] Double DTMF digits


>I wonder if your hardware is doing the actual DTMF detecting.   What 
>hardware are you using?  I'm using the  TE205P and I believe that the DTMF 
>detection is being done in the software in my case.
> Remi
>
> Steve Davies wrote:
>> On 5/3/07, Ken Leland III <k3leland at monmouth.com> wrote:
>>> When dtmfmode is set to inband for SIP, and i originate a call from sip
>>> out to the PSTN, I can hear the DTMF digit twice in the audio stream.
>>> Once very briefly and once for normal duration.
>>>
>>> Our Theory: While Asterisk is parsing the DTMF, for a fraction of a
>>> second, while the end user generated DTMF is being detected, the DTMF is
>>> passed inband. Once the DTMF is detected Asterisk silences it and
>>> regenerates it. Sensitive machines like auto attendants pick up both the
>>> brief end user generated tone as well as the full length asterisk
>>> generated tone and ultimately perceive each digit twice.
>>>
>>> Is anyone else experiencing this?
>>>
>>> I have reproduced this in an environment
>>>     * with one asterisk server that is both the feature server and the
>>> media gateway, and is timing off of network T1s
>>>     * with two servers, one feature server (timing off of ztdummy) and
>>> one media gateway (timing off of network T1s) using IAX as the inter
>>> asterisk protocol
>>>
>>> It is pretty easy to reproduce:
>>> -Dial a PSTN number(like your cell) from a sip phone using inband DTMF,
>>> and configured in asterisk sip.conf with dtmfmode=inband.
>>> -Answer the PSTN end.
>>> -Press and hold a digit on the sip phone. On the PSTN phone you will
>>> hear a very brief, end user generated, tone.
>>> -Let go of the digit on the sip phone. On the PSTN phone you will hear
>>> the asterisk generated tone.
>>>
>>> Can anyone else hear the brief initial tone?  Any help is greatly
>>> appreciated!
>>
>> Yes, we have a similar issue, but do not normally use inband DTMF
>> because SIP phones very  cleanly generate rfc2833 RTP packets directly
>> and remove this issue.
>>
>> On the other hand, asterisk is not alone dealing with this issue in
>> SIP. The Linksys ATAs have exactly the same issue.
>>
>> Strangely, I do not have a problem receiving inband DTMF through
>> Zaptel, which I believe uses the same DSP code for DTMF detection...
>> Or does it?
>>
>> Steve
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>
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