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 I doubt that I can make the necessary modifications even with your hints, but would be willing to pay for your (or anybody's) time solving the matter in a way that (a) leaves the FreePBX installation untouched as far as possible and (b) is easy to apply to subsequent versions. Ideally, it would be a slight modification of Daniel's excellent tutorial at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb, allowing to create extensions in FreePBX. Anybody willing to give it a shot? Just let me know what you would charge and we can talk about it.