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2008 Jul 19
1
going from 1.4.21 to 1.6 beta 9
1.4 was working fine.
I thought I would try 1.6 beta 9
from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept
the call.
[Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite:
Call from 'JJ' to extension 'jj_audio' rejected because extension not found.
I changed nothing in the config files.
I tried setting debug level to 5 and verbose to 5 all I still get is the
one liner above.
Has something changed in 1.6 that affects incoming calls (that I have
not found)
my sip.conf still has the context set to...