Hello,

 

the error message about the “db_DBURL not found” is most probably some syntax error or missing define on your cfg. The database modules are named db_mysql, db_postgres and so on.

 

Cheers,

 

Henning

 

From: Saul Ibarra via sr-users <sr-users@lists.kamailio.org>
Sent: Freitag, 1. Dezember 2023 21:01
To: Fred Posner <fred@pgpx.io>
Cc: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; Saul Ibarra <monkeyslippery@gmail.com>
Subject: [SR-Users] Re: Setting up Kamailio as SIP Proxy

 

Hey Fred,
I got it working thanks to your example, it is ofc not the exact same configuration i needed but it helped to understand more about the UAC module and Kamailio in general, it was my first try with it.

Related question, does an error "Module db_DBURL not found" has anything to do with the UAC module?
there is also "uac_reg_load_db(): database module not found"

For this test i used Ubuntu (because locally i was using docker in a M1 Mac and it was erroring because of the builds available on the debian registry), maybe it has anything to do with that?

You saved me man, thanks!

 

On Wed, Nov 29, 2023 at 11:06 AM Saul Ibarra <monkeyslippery@gmail.com> wrote:

Hey Fred,
thanks for the info and the example on how to handle this.

I'll take a look at it and try to get it working
Thanks again

 

On Tue, Nov 28, 2023 at 11:45 PM Fred Posner <fred@pgpx.io> wrote:

Hi Saul,

Great to see you here…

So, there’s not a ton of examples out there on UAC, but many of us have done what you’re describing and later on I’ll try to respond with a config example.

Nick vs Networking has some good, verbose discussions on many modules including;
https://nickvsnetworking.com/kamailio-bytes-sip-uac-module-to-act-as-a-uac-sip-client/

For what you’re talking about doing, I’d probably have the credentials stored in a DB (mariadb, Postgres, etc) and then have that register to the carriers.

Asterisk would connect to your Kamailio, and then using whatever logic you want (LCR, x-header, favorite, etc) you would then send out to the carrier with kamailio handling (via uac) the authentication on the INVITEs.

For incoming, it depends on the carrier how you receive the call. Some carriers will send all dids to just your registered username, others will send with DID info in the SIP.

Every carrier is a snowflake.

Keeping with the snowflake theme, some carriers are also “finicky” on outbound traffic and don’t like to see certain SIP; where some modules like topos may assist.


Regards,

Fred Posner
p: +1 (352) 664-3733



> On Nov 28, 2023, at 8:26 PM, Saul Ibarra via sr-users <sr-users@lists.kamailio.org> wrote:
>
> Thanks for your response,
> I have checked the UAC module, but i would like get more information about how it works. I have worked with Asterisk in the past, if i were to handle this let's say with an asterisk in the middle, it will probably be two sip accounts + a dialplan config to make them call each other
>
> So here the "equivalent" is UAC module to do the 'register' to the provider, but, does it means that as soon as kamailio have the provider credentials it will try to register?
>
> While using a DB (i'm using mariadb), will any update on the DB be automatically accessible by Kamailio or i should do some 'reload' for it?
>
> I'm confused by the way it works which is pretty different from Asterisk in the end, but i like the idea of having also full control over what's happening and when
>
>
> On Tue, Nov 28, 2023 at 9:39 PM Yuriy G <ovoshlook@gmail.com> wrote:
> Hi. Use UAC module
> Examples of the configuration are in the docs
>
> https://www.kamailio.org/docs/modules/5.7.x/modules/uac.html#idm700
>
> On Wed, Nov 29, 2023, 01:26 Saul Ibarra via sr-users <sr-users@lists.kamailio.org> wrote:
> Hi,
> I am needing some help setting up kamailio as a sip proxy between a provider(trunk) and a test pbx
> The trunk requires username&password and i would like to add that information to the request within the Kamailio proxy, so that the PBX can authenticate directly by IP (without username&password) through the Kamailio proxy.
>
> As a programmer with a limited background in SIP (currently learning), I am keen on understanding the intricacies of this setup at the SIP level.
> Could you please direct me to resources or documentation that covers scenarios similar to mine?
>
> Any help or guidance you could provide would be greatly appreciated.
>
> Thanks
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