On 29 May 2013 10:25, Michael Leuker <mich...@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
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