Though we’re using asterisknow, bumped to asterisk 1.6.0.15, with freePBX, the problem seems to be core asterisk/sip related.
sip_custom_post.conf,
[6002](+)
...
...
...Where the real sip extension 6002 is deleted from the rest of the dial plan, and where an #include => sip_custom_post.conf exists, and where the related [6002](+) – with emphasis on the “(+)” – is NOT deleted, this condition will prevent the entire SIP sub-system – every sip extension, every sip trunk, every-sip everything – from being loaded AT ALL.
Though it is a pilot error (mis-configuration), it should fail more gracefully.
/S