Hi Robert,<br><br>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.<br>
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.<br>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.<br>
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.<br>
<br>Cheers<br>Alex<br><br>P.S. This tutorial help me a lot <a href="http://www.kamailio.net/dokuwiki/doku.php/asterisk:realtime-integration">http://www.kamailio.net/dokuwiki/doku.php/asterisk:realtime-integration</a><br><br>
<br><br><br><br><div class="gmail_quote">On Mon, Jan 11, 2010 at 6:55 PM, Robert B <span dir="ltr"><<a href="mailto:devo@spudland.com">devo@spudland.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
<font face="Calibri">Regardless, despite my Googlage, I can't seem to
find what I am looking for.<br>
<br>
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.<br>
<br>
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,
<a href="http://voip.ms" target="_blank">voip.ms</a>, etc). I want the ACLs to be setup based on IP addresses and
not user registrations. </font><font face="Calibri">I want nothing to
do with the media or RTP streams.</font><br>
<font face="Calibri"><br>
</font><font face="Calibri">First step would be termination (outbound).
The obvious next step would be setting it up for origination (inbound).<br>
</font><br>
<font face="Calibri">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.<br>
<br>
What would the appropriate terminology for this kind of setup be? Can
anyone point out a Wiki article which I may have missed? </font><font face="Calibri">To me, this seems like the most basic of basic
configurations and a seemingly obvious launching point for anyone
getting into the software...<br>
<br>
Thanks!<br>
<br>
-- Robert<br>
<br>
</font>
</div>
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