Ian Hailey wrote:
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.[EMAIL PROTECTED] wrote:
On Fri, 22 Apr 2005, Peter Bowyer wrote:
On 22/04/05, Ian Hailey <[EMAIL PROTECTED]> 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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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