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

[Barry Flanagan]

On 29 May 2013 10:25, Michael Leuker <michael at 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 at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20130529/86a1bb12/attachment-0001.html>