I'm using sip.digiumcloud.net but eventually will be using wholesale at
bandwidth.com. I'll take a look at your links. Thanks!
On 8/21/19 10:08 AM, Karsten Horsmann wrote:
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
<[email protected] <mailto:[email protected]>>:
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.ex.config
> 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).
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> [email protected] <mailto:[email protected]>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
[email protected] <mailto:[email protected]>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Mit freundlichen Grüßen
*Karsten Horsmann*
_______________________________________________
Kamailio (SER) - Users Mailing List
[email protected]
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
[email protected]
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
