I thought there used to exist a config wizard that you could go thru and check all the options you wanted and it would spit out a working config for Kamailio... or was I dreaming? On Thu, Feb 5, 2015 at 11:44 AM, <cmrienzo@gmail.com> wrote:
The sip proxy is nice at handling the unwanted high CPS from attacks that the Sofia sip stack is incapable of dealing with. Plus the proxy can send those unwanted requests to the bit bucket instead of replying like FS will do.
Chris
On Feb 5, 2015, at 12:33, Kristian Kielhofner <kris@kriskinc.com> wrote:
Unless you're looking at extremely high scale (several hundred to thousands CPS) or tens - hundreds of thousands of call (yes, really) why not just use another FreeSWITCH instance in bypass_media? On decent hardware FreeSWITCH can easily manage a simple call routing config in bypass_media into hundreds of CPS and several thousands calls.
This can provide everything you've mentioned (and more) with many additional advantages including not needing to learn SIP in and out to make a SER-ish configuration useful.
On Mon, Feb 2, 2015 at 3:55 AM, 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
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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