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 5...@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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