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 <[EMAIL PROTECTED]> 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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