<div><div dir="auto">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.</div></div><div dir="auto">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.</div><div dir="auto">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.</div><div dir="auto"><br></div><div dir="auto">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.</div><div dir="auto"><br></div><div dir="auto">David</div><div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Tue, 21 May 2019 at 14:14, Benoit Panizzon <<a href="mailto:benoit.panizzon@imp.ch">benoit.panizzon@imp.ch</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Dear Kamailio Users<br>
<br>
I might need a bit help with starting.<br>
<br>
We are a TSP and at the moment we use a commercial carrier grade voice<br>
switch, which is very VERY inflexible and where bugs we find either<br>
don't get fixed, or only with a large delay.<br>
<br>
As everywhere else we embrace opensource and enjoy the possibility to<br>
quickly fix bugs ourselves (we have a software development department)<br>
we are evaluating Kamailio.<br>
<br>
Basically, if we have a centralized SIP routing engine, which can record<br>
CDRS for Billing and do some Number Translations (lookup on<br>
external service or database) for (location based emergency numbers,<br>
call forwarding services, ported numbers, call spam blacklisting) we<br>
could do way more than our actual $$$ carrier switch is capable of.<br>
Kamailio sounds like perfect for this task.<br>
<br>
First step:<br>
<br>
Get a machine up and running with Kamailio and Siremis, done, that was<br>
easy.<br>
<br>
Next:<br>
<br>
Connect his SIP wise via what I would call a 'SIP carrier<br>
interconnecton', ip based authenticated trunk between our carrier<br>
switch and Kamailio so we can route some number ranges from our carrier<br>
switch to Kamailio and then continue to test with subscribers,<br>
subscriber trunks and all else we need on Kamailio.<br>
<br>
I'm quite fluent in Asterisk (our Voicemail and Announcement Services<br>
are based on Asterisk) and SIP and have come across many different PBX,<br>
but after reading parts of the documentation, and trying myself to<br>
understand how Kamailio talks to other SIP endpoints I am a bit at a<br>
loss.<br>
<br>
I would have expected that I would configure our carrier switch as<br>
'remote SIP gateway' or whatever it would be called.<br>
<br>
But I start fearing, Kamilio works completely differently. So I might<br>
need a bit of pointing the right direction to get started.<br>
<br>
I did try with the 'CarrierRoute Management' or 'Dispatcher' functions,<br>
but I only find very little documentation on this.<br>
<br>
So how do I get started? How do I, for starters, tell Kamailio what the<br>
Hostname or IP of my Carrier switch is and how do I set a 'default'<br>
call route to this?<br>
<br>
Kind regards<br>
<br>
-Benoît Panizzon-<br>
-- <br>
I m p r o W a r e A G - Leiter Commerce Kunden<br>
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</blockquote></div></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>