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<body class='hmmessage'><div dir='ltr'>Check out the configs from 2600hz camp (both older opensips to their newer kamalio setup). It's not true load balancing, just round robin or linear hunting. Both opensips and kamalio have a (3rd party) gui as well. There are also cli based commands through opensipsctl and the kamailio variant to remove media servers from the pool, bring them back in, etc... The gui's helps there as well.<div><br></div><div>You will still need to dive in to docs pages for either proxy to understand what some of the config is doing as simply dumping 2600hz configs may not work for your setup, but it was a great starting point for us.<div><div><br><div><hr id="stopSpelling">From: 0x6e6562@gmail.com<br>Date: Mon, 2 Feb 2015 17:07:36 +0100<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] SIP Proxy - Kamailio/Opens* load-balancing - simple example please?<br><br><div><span></span></div><div><div>Hi Avi,</div><div><br></div><div>Albeit slightly old (nevertheless still relevant), here is a kamailio guide for freeswitch that might give you some inspiration: <a href="http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc" target="_blank">http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc</a>.</div><div><br></div><div>HTH,</div><div><br></div><div>Ben<br><br><br></div><div><br>On 2 Feb 2015, at 09:55, Avi Marcus <<a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a>> wrote:<br><br></div><blockquote><div><div dir="ltr">Hi - I know this has come up several times and I see several examples even on the old wiki - but I've never used a SIP proxy before, and I don't understand most of the configuration in the examples.<div><br></div><div>Here's what I believe is a fairly common use case:</div><div><br></div><div>1) I have inbound calls via various carriers. Some only support an IP endpoint (so DNS won't work) and some don't support 302 redirects, so I can't use a stateless FS endpoint either, so I need a sip proxy.</div><div><br></div><div>2) I need a SIP proxy that will route calls to various FreeSWITCH endpoints.</div><div>(Preferably, it should route registration too (or manage registration list itself?))</div><div><br></div><div>3) I need the ability to pull freeswitch nodes out of the routing (waiting for them to drain is fine, I don't need failover of live calls) to perform maintenance, and then add them back to the routing.</div><div><br></div><div>4) I should be able to have this proxy on a floating IP that I can move <i>this </i>too, <i>without downtime</i>, for maintenance work.</div><div><br></div><div>I think this is mostly handled by <a href="https://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs" target="_blank">the</a> <a href="http://wiki.freeswitch.org/wiki/SBC_Setup" target="_blank">examples</a> <a href="https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS" target="_blank">I</a> <a href="http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration" target="_blank">found</a> - but the opensips configuration files are hundreds of lines that I don't understand. If it's just routing calls to a backend, shouldn't that be possible in a small numbers of lines that are more understandable?</div><div><br></div><div>I imagine someone can probably just point me to a tutorial/working code that I can use. Several have been shared but I don't recall any one being particularly simple... Also, most of them tell you about compiling code. FS has been released in packages - has opensips? That might cut off many steps from an updated tutorial.</div><div><br></div><div>Thanks!</div><div><br></div><div><br></div><div>-Avi Marcus</div></div>
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