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

[Geoffrey Mina]

Matt,
How did you go about appending the ;cpd=true to your RURI.  Did you
need a patch for asterisk for that as well, or will asterisk allow you
to embed that into your Dial string?

thanks,
Geoff

On Fri, Jul 3, 2009 at 9:31 AM, Matt Florell<astmattf at gmail.com> wrote:
> Sangoma NetBorder CPD(Call Progress Detection), and we wrote a patch
> into Asterisk to be able to send the SIP messages from the CPD as AMI
> Events so that we could parse them in our application.
>
> Thanks,
>
> MATT---
>
> On 7/3/09, Venefax <venefax at gmail.com> wrote:
>> 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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