[asterisk-users] answer a voip call, play info.

[Matthew Pease]

Awesome.  Thanks for the tips.   Today I'm going to install asterisk
for the first time & try to set up the SIP/ DID trunk.  I signed up with
Voicepulse connect.

Matt

On 6/5/07, Mojo with Horan & Company, LLC <mojo at horanappraisals.com> wrote:
> I think you're on the right track.  You need to decide where to store
> the CID->data mappings (files on disk, astdb, mysql, generated
> on-the-fly) and come up with what the wave files are (text to speech?
> selected from pre-made recordings?) I would do the brunt of the work
> with a script instead of in the diaplan directly. In a fragment of a
> nutshell, and probably riddled with errors:
>
> in sip.conf,
> register => with your peer
>
> in extensions.conf,
> [incoming_context]
> exten => XXXXXXXXXX,1,Answer
> exten => XXXXXXXXXX,2,System("generate_wave_file ${CALLERID(num)}
> ${CALLERID(num)}.wav")
> exten => XXXXXXXXXX,3,Playback("${CALLERID(num)}.wav")
> exten => XXXXXXXXXX,4,Wait(3)
> exten => XXXXXXXXXX,5,goto(incoming_context,XXXXXXXXXX,3)
>
>
> Matthew Pease wrote:
> > Hi all -
> >
> > Not really sure where to post this question as I am just starting to
> > research this issue.
> >
> > We want to allow users to dial into our did voip number.   Our service
> > will:
> >
> > 1.  get their phone number via caller ID.  look up data with the caller id.
> >
> > 2.  generate a wave file based on the data returned & play it to the
> > user over the established voip link.
> >
> >
> > How might this be done using DID origination and a VOIP stack?  Or is
> > this even what I want?
> >
> > Any ideas?
> >
> >
> > Thank you--
> > Matt Pease
> > ParkingHero, Inc.
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