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

[Venefax]

I need the same functionality.
Federico

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Geoffrey Mina
Sent: Thursday, July 02, 2009 10:31 PM
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] SIP Early Media - SIT Tone Detection

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

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