Sure, here's the sequence for an inbound call via the "LPhone" trunk that was supposed to go through to extension 1001. The extension was set to "NAT" in the FreePBX settings. Just ask if you need more background.
On Wed, May 29, 2013 at 6:14 PM, Barry Flanagan <ba...@flanagan.ie> wrote: > 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 > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
Inbound -- 1001.log
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