[asterisk-dev] SIP trunk DTMF audio bleeding -- any way to suppress?

[Kirill Katsnelson]

We have a problem with DTMF tones bleeding as short tones when received 
from the trunk provider. First, there is a batch of RFC whatever the 
number is DTMF payload messages in RTP stream, and then a 30-50 ms burst 
of the matching audio tone. In other words, the provider does not mask 
out the tone completely, and allows a tail to come through. These tones 
give us some trouble.

There are multiple ways to resolve that, but I am trying to start with 
the simplest approach: the configuration. Is there a setting in Asterisk 
that would allow me to strip these audio tails? I. e. mute the channel 
and drop incoming audio (or replace with a silence) for N ms after the 
last DTMF RTP message received?

And, if not, would such a feature be of general interest?

Thanks,

  -kkm