[SR-Users] How to get started (configuring an interconnection)

[David Villasmil]

If you already setup kamailio, and if I understood what you want, take the
default cfg and uncomment the WITH_PSTN and WITH_IPAUTH defines. They’re
pretty much explained in the cfg.
Whatever you send to kamailio, it tries to find the called number in the
location table (if you enabled WITH_MYSQL), and of it doesn’t find it, it
will forward it to the defined pstn variable.
On the asterisks you need to set your kamailio as the outbound proxy. This
is all for outbound to your provider. They would need to authorize your
kamailio ip for inbound.

For inbound to your asterisks, you need to use as you said, the dispatcher
module. Let us know if you need more help on that.

David

On Tue, 21 May 2019 at 14:14, Benoit Panizzon <benoit.panizzon at imp.ch>
wrote:

> Dear Kamailio Users
>
> I might need a bit help with starting.
>
> We are a TSP and at the moment we use a commercial carrier grade voice
> switch, which is very VERY inflexible and where bugs we find either
> don't get fixed, or only with a large delay.
>
> As everywhere else we embrace opensource and enjoy the possibility to
> quickly fix bugs ourselves (we have a software development department)
> we are evaluating Kamailio.
>
> Basically, if we have a centralized SIP routing engine, which can record
> CDRS for Billing and do some Number Translations (lookup on
> external service or database) for (location based emergency numbers,
> call forwarding services, ported numbers, call spam blacklisting) we
> could do way more than our actual $$$ carrier switch is capable of.
> Kamailio sounds like perfect for this task.
>
> First step:
>
> Get a machine up and running with Kamailio and Siremis, done, that was
> easy.
>
> Next:
>
> Connect his SIP wise via what I would call a 'SIP carrier
> interconnecton', ip based authenticated trunk between our carrier
> switch and Kamailio so we can route some number ranges from our carrier
> switch to Kamailio and then continue to test with subscribers,
> subscriber trunks and all else we need on Kamailio.
>
> I'm quite fluent in Asterisk (our Voicemail and Announcement Services
> are based on Asterisk) and SIP and have come across many different PBX,
> but after reading parts of the documentation, and trying myself to
> understand how Kamailio talks to other SIP endpoints I am a bit at a
> loss.
>
> I would have expected that I would configure our carrier switch as
> 'remote SIP gateway' or whatever it would be called.
>
> But I start fearing, Kamilio works completely differently. So I might
> need a bit of pointing the right direction to get started.
>
> I did try with the 'CarrierRoute Management' or 'Dispatcher' functions,
> but I only find very little documentation on this.
>
> So how do I get started? How do I, for starters, tell Kamailio what the
> Hostname or IP of my Carrier switch is and how do I set a 'default'
> call route to this?
>
> Kind regards
>
> -Benoît Panizzon-
> --
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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