[SR-Users] sip invite proxy-authorization

[Dave & Hazel]

Thanks Daniel for your support.

Maybe my choice of language could have been better. By "dial in directly", I meant preforming a SIP INVITE without first registering with the Kamailio service (SJ-Phone speak). I am dialling in remotely via NAT and my call is being relayed successfully. I checked the address table but it's empty.

(I've modified the logging level in Kamailio.cfg and it's taken effect so I'm am working on the correct configuration file)

-----Original Message-----
From: sr-users [mailto:sr-users-bounces at lists.kamailio.org] On Behalf Of Daniel Tryba
Sent: Friday, January 19, 2018 1:56 PM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
Subject: Re: [SR-Users] sip invite proxy-authorization

On Fri, Jan 19, 2018 at 11:59:52AM -0000, Dave & Hazel wrote:
> MYSQL is enabled and the subscriber table is populated but I can dial 
> in directly without being prompted for authentication. The clue may 
> well be in the kamailio.cfg I've inherited.

This is a near default config, the AUTH route should work. There is a authentication IP whitelist in the address table, are you sending the INVITE from a whitelisted adress?

But "dial in directly" means that you can call a local extension? If so, that is allowed explicitly:

        # if caller is not local subscriber, then check if it calls
        # a local destination, otherwise deny, not an open relay here




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