Source DTMF Leaking to SIP

Hi All,

We run a smallish call centre using Asterisk and we have a DTMF detecting system between the headsets and the phones that plugs into the PC’s and types in the card numbers like a virtual keyboard (For CC Processing).

We’re having issues with very small snippets of the DTMF tone from the source line being played on the handset as well as the tone injected on the Asterisk server, are there any settings to adjust the sensitivity of the DTMF detection and stop the original source tones making it to the handsets?

Thanks a lot for any help/advice anyone can offer.

Asterisk doesn’t deliberately delay the through audio. The only way of suppressing the start of the tone completely would be to delay the audio by the time it takes to recognize the tones (the maths of which mean that it will always take a non-trivial amount of time) and then deleting the delayed audio from before the detection point.

Also, unless you are using dahdi for your incoming leg, it is probable that the DTMF detection and muting is being done before it reaches Asterisk.

Thanks for the replies, yes I’m using dahdi for incoming. Back to the drawing board by the sounds of it then as I’m always likely to get some noise from the original dtmf tone come through.