[asterisk-dev] SIP Early Media - SIT Tone Detection

[Geoffrey Mina]

Hello,
I am looking for a developer who may be interested in developing SIT
Tone Detection functionality into chan_sip. Most of my carriers do not
return disconnects as SIP error codes, instead they simply send 100
trying, followed by early media which would have the tri-tone followed
by a message that the number is invalid.

I have a need to have the dialresult properly set to INTERCEPT (or
similar) if Asterisk eventually cancels the INVITE.  This is a
scenario that currently results in a NOANSWER.

If anyone is interested in taking on this work (for $$ obviously)
please let me know.

Thanks,
Geoff