Hi all -
Not really sure where to post this question as I am just starting to research this issue.
We'd like an all VOIP solution where we have a telephone number that terminates at our server.
At our server, we'd like to:
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 is this done? Totally new to the game here.
I've read about DID origination, SIP channels, SIP peers... its all quite confusing!
I'm a java developer, so something that works via Java would be great, but scripting in various languages is no big deal. The application doesn't need to do much more than query a database & choose a WAV file to play back based on that info.
Thank you-- Matt Pease ParkingHero, Inc.
Your answers are on http://www.voip-info.org/wiki/ (a site with planty of VoIP related informations), http://www.asterisk.org (an IP-PBX with many, many, many features) and, of course, the OpenSER wiki site (http://www.openser.org/dokuwiki/doku.php).
Edson
Hi Edson -- I've seen those sites. However, there is too much information there & what I need to do seems simple & usual. I haven't found a great FAQ that answers these basic questions. So I thought someone out there might be able to get me bootstrapped.
Thanks - Matt
On 6/4/07, Edson 4lists@gmail.com wrote:
So You come up with the Holly about VoIP... There is not a simple answer... You could use Asterisk (which I think would Your best choise) or a combination of *SER and SEMs, or any other kind of implementation that could combine signalization and RTP management. Other examples could also be finded on Yate solution.
So, in a short answer, You have to dig Yourself on one of this sites, study there capabilities and find out which one best suit to Your needs.
Sorry, but as I said, there is no a simple answer to Your question. That's not simple to follow all the related RFCs to make a 'simple answer machine'.
Good look in Your study.
If You have more specific questions about how OpenSER works and what is necessary to make it work, after reading the Wiki material, post Your questions on this forum... We'll be glad to help You.
Edson
Hello,
Matthew Pease wrote:
If python is ok for you, SEMS' ivr application might be useful to you. Have a look at the python part of the app tutorial (http://www.iptel.org/sems/sems_application_development_tutorial), and Juha's db based announcement application: http://lists.iptel.org/pipermail/semsdev/2007-May/001477.html where you will probably replace self.dialog.user with some part of self.dialog.remote_uri or self.dialog.remote_party to have the announcement dependent on caller and not called party.
Stefan
Hello Matthew,
as OpenSER handly mostly the SIP part of the call, you'll need some other solution for the audio playback, e.g. Asterix.
There is the seas module in OpenSER where you can program your application in java, you can use also the perl module for scripting inside the server.
But for simple playback just look into asterix, you can script it with perl too.
Cheers,
Henning
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Edson -
henning Westerholt -
Matthew Pease -
Stefan Sayer