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