Hi Avi, Albeit slightly old (nevertheless still relevant), here is a kamailio guide for freeswitch that might give you some inspiration: http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc. HTH, Ben
On 2 Feb 2015, at 09:55, Avi Marcus <avi@avimarcus.net> wrote:
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 examples I found - 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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