Kamailio can do this. Asterisk also can do this, and for small number (less then few thousands of clients, may be Asterisk is even better - my personal opinion ).<div>You can install kamailio, and put its default conf file and - you are ready to start tweaking. If Kamailio have to register to other sip server as client - I am not sure it is possible, but Asterisk can do it easy.</div>
<div>Everything else is easy. Kamailio will call users by their respective address - and you will not care how it will look for IP or interface. Or you can make some routing in its conf file and that's all.</div><div>
You will need to read, then to read, again you will need reading, testing etc.</div><div>Open source products are free. You can get them and use them in a way you wish for free. But you must know how to use them, and this is not that easy.</div>
<div>I spent around 2 months to make what I need. And at the end, I fixed some problems dirty way, because I dont know better way.</div><div><br><br><div class="gmail_quote">On Mon, Jan 30, 2012 at 7:00 PM, Me <span dir="ltr"><<a href="mailto:mojo1736@privatedemail.net">mojo1736@privatedemail.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Apologies if any of the questions below are a bit dumb - I don't pretend to be an expert in SIP/VOIP - I am just an ordinary user looking for answers.<br>
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Our current setup involves processing a small number of internal sip accounts (up to 10, no more than that) and one "public" one (with a separate registrar) in the following way:<br>
<br>
On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our entry point to the public internet, eth1 faces our internal network and tun0 is a private vpn, which connects all our smartphones to the internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility.<br>
<br>
Up until now, we have been routing voip calls via a commercial (closed source), very limited, terribly outdated (Pentium code base!) and rather buggy sip proxy. I had to employ a lot of hacks on our server in order to route calls as this proxy can only listen on a single interface. It was also a nightmare to maintain. Unsurprisingly, I decided that enough is enough and I am now determined to replace it.<br>
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We route calls in the following way: all machines (PCs are all Linux based) & smartphones have their own sip/voip client installed on them (also using bluetooth). Internal calls are routed via the proxy between ourselves either on the internal net (eth1), or between the vpn and eth1 (eth1<->tun0).<br>
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External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0.<br>
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As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues:<br>
<br>
I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way:<br>
- calls made to <userX>@<a href="http://ourdomain.net" target="_blank">ourdomain.net</a> to be routed internally via eth1 (internal net) or tun0 (private vpn);<br>
- calls made to anybody else to be routed externally via eth0 (public) using the separate "public" sip account with our external registrar;<br>
- calls made to the public sip account (from outside - the "public") need to be routed to a "nominated" internall account (say <user0>@<a href="http://ourdomain.net" target="_blank">ourdomain.net</a>);<br>
- all other (internal) calls need to be routed depending on which interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones).<br>
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Obviously, calls need to be received (and routed properly) from all 3 interfaces.<br>
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Is all of this possible with Kamailio?<br>
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I want to avoid unnecessary complexities of the setup (as I already mentioned above - I am just a user and by no means an expert in sip/voip) and do not want to deploy something I do not need - I need to keep the memory footprint to a bare minimum, possibly without sacrificing performance.<br>
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Once this is done, I would then move on to the next phase and use IM & ENUM, but this is once the above works.<br>
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I looked at other alternatives, but I got very confused there as well - I couldn't figure out what exactly is the difference between, say, OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup?<br>
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One last thing - I am a developer by trade and I am not afraid of "tweaking" things when needed. I was successful in compiling Kamailio from source (I use Fedora on all our machines) and I was pleased that I could exclude from the RPM .spec file the modules I think I did not need.<br>
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I also made some modification of my own to make the database modules (mysql, postgresql and unixodbc) configurable in the same way the rest of the modules are. I could submit patches, if needed, so that these are incorporated into future releases - how do I do that?<br>
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I could not do the same with OpenSIPS, however (which I also tried - out of curiosity!) - everything there seems to be lumped and compiled together regardless of whether it is needed or not.<br>
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Any help as to helping me with the above issues is greatly appreciated, many thanks in advance for taking the time!<br>
<br>
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