<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">There is no way to make a sip user agent stateless. recovery may be an option.<div class=""><br class=""><div><blockquote type="cite" class=""><div class="">On Feb 5, 2015, at 1:59 PM, Avi Marcus <<a href="mailto:avi@avimarcus.net" class="">avi@avimarcus.net</a>> wrote:</div><br class="Apple-interchange-newline"><div class=""><div dir="ltr" class="">Kristian: Indeed, that would be easier! <div class="">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 class=""><br class=""></div><div class="">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" class=""><div class=""><div class="gmail_signature"><div dir="ltr" class="">-Avi<br class=""></div></div></div>
<br class=""><div class="gmail_quote">On Thu, Feb 5, 2015 at 7:33 PM, Kristian Kielhofner <span dir="ltr" class=""><<a href="mailto:kris@kriskinc.com" target="_blank" class="">kris@kriskinc.com</a>></span> wrote:<br class=""><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 class="">
thousands CPS) or tens - hundreds of thousands of call (yes, really)<br class="">
why not just use another FreeSWITCH instance in bypass_media? On<br class="">
decent hardware FreeSWITCH can easily manage a simple call routing<br class="">
config in bypass_media into hundreds of CPS and several thousands<br class="">
calls.<br class="">
<br class="">
This can provide everything you've mentioned (and more) with many<br class="">
additional advantages including not needing to learn SIP in and out to<br class="">
make a SER-ish configuration useful.<br class="">
<span class="im HOEnZb"><br class="">
On Mon, Feb 2, 2015 at 3:55 AM, Avi Marcus <<a href="mailto:avi@avimarcus.net" class="">avi@avimarcus.net</a>> wrote:<br class="">
</span><div class="HOEnZb"><div class="h5">> Hi - I know this has come up several times and I see several examples even<br class="">
> on the old wiki - but I've never used a SIP proxy before, and I don't<br class="">
> understand most of the configuration in the examples.<br class="">
><br class="">
> Here's what I believe is a fairly common use case:<br class="">
><br class="">
> 1) I have inbound calls via various carriers. Some only support an IP<br class="">
> endpoint (so DNS won't work) and some don't support 302 redirects, so I<br class="">
> can't use a stateless FS endpoint either, so I need a sip proxy.<br class="">
><br class="">
> 2) I need a SIP proxy that will route calls to various FreeSWITCH endpoints.<br class="">
> (Preferably, it should route registration too (or manage registration list<br class="">
> itself?))<br class="">
><br class="">
> 3) I need the ability to pull freeswitch nodes out of the routing (waiting<br class="">
> for them to drain is fine, I don't need failover of live calls) to perform<br class="">
> maintenance, and then add them back to the routing.<br class="">
><br class="">
> 4) I should be able to have this proxy on a floating IP that I can move this<br class="">
> too, without downtime, for maintenance work.<br class="">
><br class="">
> I think this is mostly handled by the examples I found - but the opensips<br class="">
> configuration files are hundreds of lines that I don't understand. If it's<br class="">
> just routing calls to a backend, shouldn't that be possible in a small<br class="">
> numbers of lines that are more understandable?<br class="">
><br class="">
> I imagine someone can probably just point me to a tutorial/working code that<br class="">
> I can use. Several have been shared but I don't recall any one being<br class="">
> particularly simple... Also, most of them tell you about compiling code. FS<br class="">
> has been released in packages - has opensips? That might cut off many steps<br class="">
> from an updated tutorial.<br class="">
><br class="">
> </div></div></blockquote></div></div></div></div></blockquote></div></div></body></html>