4 Jun
2007
4 Jun
'07
11:40 p.m.
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.
5 Jun
5 Jun
1:08 a.m.
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
-----Original Message-----
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
Behalf Of Matthew Pease
Sent: segunda-feira, 4 de junho de 2007 17:41
To: users(a)openser.org
Subject: [Users] application via voip?
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.
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
1:22 a.m.
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(a)gmail.com> wrote:
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
-----Original Message-----
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
Behalf Of Matthew Pease
Sent: segunda-feira, 4 de junho de 2007 17:41
To: users(a)openser.org
Subject: [Users] application via voip?
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.
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
2:48 a.m.
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
-----Original Message-----
From: Matthew Pease [mailto:matt@parkinghero.com]
Sent: segunda-feira, 4 de junho de 2007 19:23
To: Edson
Cc: users(a)openser.org
Subject: Re: [Users] application via voip?
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(a)gmail.com> wrote:
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
>-----Original Message-----
>From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
>Behalf Of Matthew Pease
>Sent: segunda-feira, 4 de junho de 2007 17:41
>To: users(a)openser.org
>Subject: [Users] application via voip?
>
>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.
> >
> >_______________________________________________
> >Users mailing list
> >Users(a)openser.org
> >http://openser.org/cgi-bin/mailman/listinfo/users
>
> 
3:55 a.m.
Hello,
Matthew Pease wrote:
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.
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
Thank you--
Matt Pease
ParkingHero, Inc.
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
--
Stefan Sayer
Media Services Development
stefan.sayer(a)iptego.de
www.iptego.de
iptego GmbH
Am Borsigturm 40
13507 Berlin
Germany
Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
6 Jun
6 Jun
8:30 a.m.
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.
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.
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.
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
6379
days inactive
6381
days old
5 comments
4 participants
participants (4)
-
Edson -
henning Westerholt -
Matthew Pease -
Stefan Sayer