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. Here's what I believe is a fairly common use case: 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. 2) I need a SIP proxy that will route calls to various FreeSWITCH endpoints. (Preferably, it should route registration too (or manage registration list itself?)) 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. 4) I should be able to have this proxy on a floating IP that I can move *this *too, *without downtime*, for maintenance work. I think this is mostly handled by the <https://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs> examples <http://wiki.freeswitch.org/wiki/SBC_Setup> I <https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> found <http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration> - 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? 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. Thanks! -Avi Marcus