Hi,
I've a few working asterisk servers, all seeing the same symptom, but
they are all based on the same configs.
A SIP inbound INVITE message is coming in to an extension (not a peer)
eg 555 at ourserver.com
A tcpdump clearly shows the INVITE coming in, but asterisk seems to be
ignoring it (theres no reply outbound packet). All the source/dest IPs
and ports look good.
A "sip set debug trace ip <sourceip>" is blank, showing nothing
at all.
The sip.conf default context is incoming_pstn. The incoming_pstn context
is:
[incomming_pstn]
include => local-UK
include => local-US
include => test_numbers
and [test_numbers] includes:
exten => 555,1,Answer(0) ; Pick up phone instantly
exten => 555,n,Playback(vq51) ; Let them know what's going on
exten => 555,n,Playback(vq20)
exten => 555,n,Goto(default,555,3) ; repeat
So as far as I can tell, we should be accepting the connection and
playing the voicefile (yup - I know this would be open to the internet,
that's the intention).
Sip.conf also has:
allowexternalinvites=yes
allowexternaldomains=yes
so it should be working I think...
This is a 1.4.15 based asterisk
Thanks
Adrian
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