[SR-Users] Freepbx 2.11.0rc1 with Asterisk 11.3.0 and Kamailio 4.0.1

[Michael Leuker]

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 at 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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130529/61006618/attachment.html>