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

[Kirill Katsnelson]

On 110727 0134, Benny Amorsen wrote:
> Kirill Katsnelson<kkm at adaptiveai.com>  writes:
>> And, if not, would such a feature be of general interest?
>
> it would be difficult to apply to
> DTMF in SIP INFO, and it would be tricky to do it with fast-switched
> or reinvited RTP.

You are so right. There are so many ways DTMF can be handled, that 
probably it would be too hard to implement it cleanly and in a generic 
manner. So, I am backing off on the idea.

Thanks for sharing your thoughts!

  -kkm