Hi Thomas, Welcome to the SIP club ;) - we all the time looking for new victims ;) .
What your partner told you is right - the typical approach is to front end a cluster of SIP boxes ( PBX, servers, GWs, etc) with opensips for multiple purposes: security, failover, load-balancing, etc.
Now, depending what is the distribution logic you may use different functionalities in opensips. Like if you wan to simply distribute the calls over the FS boxes, try the load balancer functionality (see http://www.opensips.org/Resources/DocsTutLoadbalancing) - dispatching works in a similar way.
For dialed number based routing (like in your case), you can use Dynamic Routing module (prefix based routing) or the dialplan module (regexp based detection of numbers and routing). DR has a built in failover mechanism, for dialplan you need to do it manually from script - but first read the docs for these 2 modules.
At the end, yes OpenSIPS is the right tool for that - as version use the 1.7.2 (latest current stable).
Regards, Bogdan On 03/21/2012 11:29 AM, Thomas Løcke wrote:
Hey all, Let me start by saying that I'm a complete beginner in the arts of SIP, but I'm eager to learn, and usually not too thick to understand fairly complicated things. :o) I've become involved in a project that at some point is going to need a few FreeSWITCH servers, so naturally we've started learning how to configure, maintain and use FreeSWITCH. Then the other day I had a meeting with a potential business partner (a telco using Asterisk and OpenSIPS), and he told me about the wonders of OpenSIPS, and urged me to add OpenSIPS to our stack, since it would, and I quote: "greatly simplify our setup, while giving us more control". His main point was that instead of having many FreeSWITCH servers facing the world, I could instead settle on a few OpenSIPS servers forwarding calls to the relevant FreeSWITCH server(s), depending on the number called. This sorta/kinda made sense to me, as I'm already doing something similar with HTTP proxies: Many webservers hidden behind a few HTTP proxies. Where things start to fall apart for me, is figuring out if OpenSIPS is the tool for this job. When looking at the website and the documentation, OpenSIPS appears to me to do a whole lot more than "just proxying", and coming at it as a beginner seems like a very daunting task. By the way, I have ordered a bunch of SIP related books from Amazon (and even an OpenSIPS book!), so I should have plenty of reading material readily available soon. What I need OpenSIPS to do is route calls to specific PSTN numbers to specific FreeSWITCH servers. Lets say I have 1000 PSTN numbers and 10 FreeSWITCH servers. When my telco receives a call on numbers 1-100 I want OpenSIPS to route the call to FreeSWITCH server 1. Calls to PSTN number 101-200 goes to FreeSWITCH server 2 and so on and so forth. At some point we might also want to add some fail over and load balancing logic to the setup. Is this something that can be done using OpenSIPS? Is it the right tool for the job? Or should I perhaps redefine the job completely? My next question is: What OpenSIPS version? > From looking at the docs, it seems to me that version 2 of OpenSIPS can actually do what I need, and seeing as this project will not be brought to market anytime soon, maybe we could just as well aim for version 2 instead of going with the "old" 1.7.x branch? Or am I missing something? Any and all advice (and reading material!) are more than welcome. Regards, Thomas Løcke _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
-- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
