<div dir="ltr">Kristian: Indeed, that would be easier! <div>However, that doesn't seem to solve the ability to do maintenance on that box (I was planning to have opensips share with one of the FS instances).</div><div><br></div><div>Is recover call table from a shared sql DB reliable for that situation, or any other way to make it stateless...?</div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature"><div dir="ltr">-Avi<br></div></div></div>
<br><div class="gmail_quote">On Thu, Feb 5, 2015 at 7:33 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Unless you're looking at extremely high scale (several hundred to<br>
thousands CPS) or tens - hundreds of thousands of call (yes, really)<br>
why not just use another FreeSWITCH instance in bypass_media? On<br>
decent hardware FreeSWITCH can easily manage a simple call routing<br>
config in bypass_media into hundreds of CPS and several thousands<br>
calls.<br>
<br>
This can provide everything you've mentioned (and more) with many<br>
additional advantages including not needing to learn SIP in and out to<br>
make a SER-ish configuration useful.<br>
<span class="im HOEnZb"><br>
On Mon, Feb 2, 2015 at 3:55 AM, Avi Marcus <<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>> wrote:<br>
</span><div class="HOEnZb"><div class="h5">> Hi - I know this has come up several times and I see several examples even<br>
> on the old wiki - but I've never used a SIP proxy before, and I don't<br>
> understand most of the configuration in the examples.<br>
><br>
> Here's what I believe is a fairly common use case:<br>
><br>
> 1) I have inbound calls via various carriers. Some only support an IP<br>
> endpoint (so DNS won't work) and some don't support 302 redirects, so I<br>
> can't use a stateless FS endpoint either, so I need a sip proxy.<br>
><br>
> 2) I need a SIP proxy that will route calls to various FreeSWITCH endpoints.<br>
> (Preferably, it should route registration too (or manage registration list<br>
> itself?))<br>
><br>
> 3) I need the ability to pull freeswitch nodes out of the routing (waiting<br>
> for them to drain is fine, I don't need failover of live calls) to perform<br>
> maintenance, and then add them back to the routing.<br>
><br>
> 4) I should be able to have this proxy on a floating IP that I can move this<br>
> too, without downtime, for maintenance work.<br>
><br>
> I think this is mostly handled by the examples I found - but the opensips<br>
> configuration files are hundreds of lines that I don't understand. If it's<br>
> just routing calls to a backend, shouldn't that be possible in a small<br>
> numbers of lines that are more understandable?<br>
><br>
> I imagine someone can probably just point me to a tutorial/working code that<br>
> I can use. Several have been shared but I don't recall any one being<br>
> particularly simple... Also, most of them tell you about compiling code. FS<br>
> has been released in packages - has opensips? That might cut off many steps<br>
> from an updated tutorial.<br>
><br>
> Thanks!<br>
><br>
><br>
> -Avi Marcus<br>
><br>
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</span><span class="HOEnZb"><font color="#888888">--<br>
Kristian Kielhofner<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
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