Hi Travis,
two projects that enables siptrunking and routing:
https://github.com/voiceboys/sbcOS (ip-auth based or registrar on your side) https://dsiprouter.readthedocs.io/en/latest/
Also intressting to see how they solved this problems.
If you could describe your siptrunk a bit more, then would here many people that can point you in the right direction how to solve that with or without kamailio.
Cheers Karsten
Am Di., 20. Aug. 2019 um 17:11 Uhr schrieb Travis Ryan < travis@travisryan.com>:
Thanks,
I want to eventually get to a setup like the one here: https://github.com/CyCoreSystems/asterisk-k8s-demo
But since I'll need Kamailio to handle a high load of incoming calls, I think I need it to direct traffic, etc for any number of Asterisk servers behind it.
In this setup it indeed has RTPProxy, etc. I just want to understand how to use it rather than just drop it in, etc. Also the demo doesn't have any config for an outside SIP trunk, etc.
Maybe this helps?
Thanks, Travis
On 8/20/19 11:01 AM, Daniel Tryba wrote:
On Tue, Aug 20, 2019 at 10:22:26AM -0400, Travis Ryan wrote:
What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need
to
still register the trunk from each Asterisk box "thru" the Kamailio
proxy,
etc?
Also, I'm merely accepting outside calls and then validating the caller
and
bridging them back out to the PSTN, so I don't have any local SIP
clients,
etc., so no need to register the sip devices, etc.
The real question is what do you need kamailio to do? You answer this with as a simple proxy.
A possible solution for you is to use kamilio with the dispatcher
module. One
id (1) for the PSTN side, one id (2) for the Asterisk side. If a call
comes in
from 1, route it to 2 and v.v.
This makes the kamailio machine the "endpoint" for both PSTN and Asterisk side.
With the "default" config that comes with kamailio all you need to do is strip out anything from the accounting bit in request_route (line 508) https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg and insert something like:
if(ds_is_from_list("1",3)) { $avp(dispatcherid)="2"; } else if(ds_is_from_list("2",3)) { $avp(dispatcherid)="1"; } else { send_reply("403", "Go away"); exit; }
route(DISPATCHER); route(RELAY);
With route DISPATCHER being: route[DISPATCHER] { if(!ds_select_dst($avp(dispatcherid), "4")) { send_reply("501", "No dispatcher available"); exit; }
t_on_failure("RTF_DISPATCH"); return;}
See
https://kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher....
for more info on integrating the dispatcher module.
More advanced subjects to look at are: -do you need an rtp proxy? -do you need topology hiding? -is NAT involved?
But leave them until you have a clue about how to use kamailio as a sip
proxy in a
simple test environment (e.g. between 2 asterisk servers).
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