Hi list!
Very strange...
I ran the Asterisk CLI for other tasks, and suddenly I got this message:
== Using SIP RTP CoS mark 5
-- Executing [000972592603325@default:1]
Verbose("SIP/192.168.20.120-0000002a", "2,PROXY Call from 0123456 to
000972592603325") in new stack
== PROXY Call from 0123456 to 000972592603325
-- Executing [000972592603325@default:2] Set("SIP/192.168.20.120-0000002a",
"CHANNEL(musicclass)=default") in new stack
-- Executing [000972592603325@default:3]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialluca") in new stack
-- Executing [000972592603325@default:4]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialfax") in new stack
-- Executing [000972592603325@default:5]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialanika") in new stack
-- Executing [000972592603325@default:6]
Dial("SIP/192.168.20.120-0000002a", "SIP/pbxluca/000972592603325,,R") in new
stack
[Jun 8 21:42:50] WARNING[18981]: app_dial.c:2345 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [000972592603325@default:7]
Hangup("SIP/192.168.20.120-0000002a", "") in new stack
== Spawn extension (default, 000972592603325, 7) exited non-zero on
'SIP/192.168.20.120-0000002a'
[Jun 8 21:43:22] WARNING[16633]: chan_sip.c:3830 retrans_pkt: Retransmission
timeout reached on transmission 8dc31ca4e660a0408450715638784d86 for seqno 1
(Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
At the time no phone try to call...
On my Firewall I see a SIP packet coming from an IP in Palestine...
Am I cracked? I think I disabled all "guest" access. How can I check if my
Asterisk allows guest to originate calls?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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