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

[Venefax]

What is the name of that "proprietary solution"??

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Matt Florell
Sent: Friday, July 03, 2009 8:55 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] SIP Early Media - SIT Tone Detection

On 7/3/09, Steve Totaro <stotaro at asteriskhelpdesk.com> wrote:
> On Fri, Jul 3, 2009 at 5:15 AM, Venefax<venefax at gmail.com> wrote:
>  > 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
>  >
>
>
> Not saying that he has this or has any interest in this, but you could
>  contact Justin Newman.  He wrote NVFaxdetect or one of the or maybe it
>  was an AMD app, too lazy and early to look up.
>
>  He also wrote a cool app that could detect your gender to a pretty
>  good degree of accuracy.
>
>  I wouldn't be surprised if he already has this or could churn out
>  something fairly quickly, if interested.
>
> --
>  Thanks,
>  Steve Totaro

All of Justin's scripts operate post-Answer signal, SIT tones and
other early media unfortunately need to be analyzed while the Dial is
still occuring, making it all a lot more difficult to write into
Asterisk. We pretty much gave up on doing this ourselves and found a
proprietary solution that works very well.

Thanks,

MATT---

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