[Asterisk-Users] DTFM tones almost completly muted.

[Ian Hailey]

Ian Hailey wrote:

> steve at daviesfam.org wrote:
>
>> On Fri, 22 Apr 2005, Peter Bowyer wrote:
>>
>>  
>>
>>> On 22/04/05, Ian Hailey <asterisk at dinplug.com> wrote:
>>>   
>>>
>>>> Hello everyone,
>>>>
>>>> I am trying to receive DTMF commands on asterisk from PSTN calls
>>>> terminated at my asterisk box. I have tried to terminate the PSTN 
>>>> calls
>>>> with both SIP and IAX using sigate.co.uk and voipuser as the PSTN
>>>> terminator. When I listen to tones sent from the PSTN side (e.g.
>>>> continuous DTMF tone of about 3 seconds) on the asterisk server 
>>>> (stored
>>>> in the voice mail) the tone is more or less completely muted, just the
>>>> initial tone start can be heard. I am using the G711 codec. Does 
>>>> anyone
>>>> have any idea if these tones are on purpose muted by the service
>>>> providers or any other reason why it does not work?
>>>>     
>>>
>>
>>
>> Most likely the DTMF tones have been detected at the point where the 
>> call was converted PSTN->SIP/IAX, and forwarded instead as an 
>> indication (ie via SIP INFO or RFC2833 or whatever.  So you won't 
>> hear them in a recording of the audio stream.  The remaining blip is 
>> just the little bit at the start before the gateway recognised the tone.
>>
>> You should receive the indication in your SIP or IAX connection and 
>> Asterisk should see it (but its not audio any more).
>>
>> Regards,
>> Steve
>>
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>>  
>>
> Hi Steve,
>
> Good point, it makes sense that this is what is happening and most 
> likely at the PSTN termination point. The question is where has the 
> signalling gone as I seem not to receive it at my asterisk server. Do 
> you think that this is a configuration problem at the PSTN terminators 
> site or do they do this on purpose so they can charge extra for the 
> information etc?
>
> Thanks.
>
> Ian Hailey.
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>
OK I found that it does work correctly with PSTN-IAX termination from 
voipuser.co.uk for example so it is realy a problem with sipgate.