On 29 May 2013 10:25, Michael Leuker <michael@leuker.me> wrote:
Thank you very much for sharing your insights, Barry! I am facing the same problem that Trevor described:

Things are working just fine on their own, but as soon as FreePBX comes into play, calling extensions becomes impossible because of the different tables used. Removing the password from FreePBX (and setting the Kamailio IP in the ACL field) seems to mitigate the issue somewhat, but even though the extension shows as registered in FreePBX, it always shows as busy:

chan_sip.c:23237 handle_response_invite: Failed to authenticate on INVITE to '"xxxxxxxx" <sip:xxxxxxxx@198.23.139.21>;tag=as72a4117a'
    -- SIP/1001-00000006 is circuit-busy


Can you do "sip set debug on" on Asterisk and make a call and  post the output?

-Barry