11 Jan
2010
11 Jan
'10
11:55 a.m.
Regardless, despite my Googlage, I can't seem to find what I am looking for.
I've looked at the Wiki and the documentation. While I do have a
programming background, I still am having a hard time grasping some of
these tutorials. Many of them seem very specific and not at all relevant
to what I am trying to do. I feel like I'm left without a good starting
point.
My goal is to have Kamailio setup as an ITSP would. That is, I have
three sipxecs PBX systems here, and rather than have each one configured
with umpteen SIP trunks that I'll need to manage, I want to have (for
the time being) just one -- that one would be my personal Kamailio
server. When a user makes a call, it first goes to my Kamailio server
which then distribute the calls to my ITSPs (like Bandwidth.com,
voip.ms, etc). I want the ACLs to be setup based on IP addresses and not
user registrations. I want nothing to do with the media or RTP streams.
First step would be termination (outbound). The obvious next step would
be setting it up for origination (inbound).
I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but
as I said my lack of understanding of the terminology's context
prohibits me from really exploring anything. Some areas look familiar,
others seem redundant, etc. The reason I started with the web UI was
because it's obviously easier and it also produces output that is
syntactically correct for my own examination and learning purposes.
What would the appropriate terminology for this kind of setup be? Can
anyone point out a Wiki article which I may have missed? To me, this
seems like the most basic of basic configurations and a seemingly
obvious launching point for anyone getting into the software...
Thanks!
-- Robert
12 Jan
12 Jan
3:12 a.m.
Hi Robert,
Kamailio is perfect for what you want to do. I'm a new user also with a
litle programming skills and in the begining seems to be difficult but after
a few days you will see that is not so hard. If i was you I would prefer to
learn how the script works than learning the seremis. In this way you will
have more understanding on how things work.
So your task is to get all the calls from your 3 PBXs and send them to
kamailio. There some routing is done and the calls are send to your
Interconnections.
You can authenticate your PBXs by IP or with user/pass, depends what you
need.You can check by IP with trusted module and then send the calls
accordinly or you can use the dialplan module or the carrieroute module.
Kamailio is very very flexible and you can do anything you want.
Before you do all that is good if you setup kamailio with 2 sip phones and
play with them (authentication,route calls by request username $rU and many
other things). When you complete those then you will be able to understand
how to acomplish what you need.
Cheers
Alex
P.S. This tutorial help me a lot
http://www.kamailio.net/dokuwiki/doku.php/asterisk:realtime-integration
On Mon, Jan 11, 2010 at 6:55 PM, Robert B <devo(a)spudland.com> wrote:
Regardless, despite my Googlage, I can't seem to
find what I am looking
for.
I've looked at the Wiki and the documentation. While I do have a
programming background, I still am having a hard time grasping some of these
tutorials. Many of them seem very specific and not at all relevant to what I
am trying to do. I feel like I'm left without a good starting point.
My goal is to have Kamailio setup as an ITSP would. That is, I have three
sipxecs PBX systems here, and rather than have each one configured with
umpteen SIP trunks that I'll need to manage, I want to have (for the time
being) just one -- that one would be my personal Kamailio server. When a
user makes a call, it first goes to my Kamailio server which then distribute
the calls to my ITSPs (like Bandwidth.com, voip.ms, etc). I want the ACLs
to be setup based on IP addresses and not user registrations. I want
nothing to do with the media or RTP streams.
First step would be termination (outbound). The obvious next step would be
setting it up for origination (inbound).
I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x, but as
I said my lack of understanding of the terminology's context prohibits me
from really exploring anything. Some areas look familiar, others seem
redundant, etc. The reason I started with the web UI was because it's
obviously easier and it also produces output that is syntactically correct
for my own examination and learning purposes.
What would the appropriate terminology for this kind of setup be? Can
anyone point out a Wiki article which I may have missed? To me, this seems
like the most basic of basic configurations and a seemingly obvious
launching point for anyone getting into the software...
Thanks!
-- Robert
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
6:27 a.m.
New subject: [Kamailio-Users] This has probably been asked a thousand times before...
Hello,
first a suggestion - please use relevant subject message for what you
look for, it helps when checking the archive and looking to get same
information.
Now, if you are going to use the latest stable release 3.0.0 from yesterday:
http://www.kamailio.org/mos/view/Kamailio-OpenSER-v3.0.0-Release-Notes
then you can read the getting started document for SER - since Kamailio
(OpenSER) and SER work together in http://sip-router.org project, and
version 3.0.0 is a result of that work, the getting started is good to
start with. It is a bit obsoleted in terms of what you can do now with
latest version but you can get the idea of architecture and cfg structure:
http://siprouter.teigre.com/doc/gettingstarted/
From there, wiki and mailing lists can help you easier.
Hope it helps.
Cheers,
Daniel
On 1/11/10 5:55 PM, Robert B wrote:
Regardless, despite my Googlage, I can't seem to
find what I am
looking for.
I've looked at the Wiki and the documentation. While I do have a
programming background, I still am having a hard time grasping some of
these tutorials. Many of them seem very specific and not at all
relevant to what I am trying to do. I feel like I'm left without a
good starting point.
My goal is to have Kamailio setup as an ITSP would. That is, I have
three sipxecs PBX systems here, and rather than have each one
configured with umpteen SIP trunks that I'll need to manage, I want to
have (for the time being) just one -- that one would be my personal
Kamailio server. When a user makes a call, it first goes to my
Kamailio server which then distribute the calls to my ITSPs (like
Bandwidth.com, voip.ms, etc). I want the ACLs to be setup based on IP
addresses and not user registrations. I want nothing to do with the
media or RTP streams.
First step would be termination (outbound). The obvious next step
would be setting it up for origination (inbound).
I've downloaded and installed the SIREMIS web UI for OpenSER 1.4.x,
but as I said my lack of understanding of the terminology's context
prohibits me from really exploring anything. Some areas look familiar,
others seem redundant, etc. The reason I started with the web UI was
because it's obviously easier and it also produces output that is
syntactically correct for my own examination and learning purposes.
What would the appropriate terminology for this kind of setup be? Can
anyone point out a Wiki article which I may have missed? To me, this
seems like the most basic of basic configurations and a seemingly
obvious launching point for anyone getting into the software...
Thanks!
-- Robert
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
* http://www.asipto.com/
5473
days inactive
5474
days old
2 comments
3 participants
participants (3)
-
alex pappas -
Daniel-Constantin Mierla -
Robert B