[asterisk-users] Am I cracked?

Luca Bertoncello
2015-06-08T19:46:54Z

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 at 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 at default:2]
Set("SIP/192.168.20.120-0000002a",
"CHANNEL(musicclass)=default") in new stack
    -- Executing [000972592603325 at default:3]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialluca") in new
stack
    -- Executing [000972592603325 at default:4]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialfax") in new
stack
    -- Executing [000972592603325 at default:5]
GotoIf("SIP/192.168.20.120-0000002a", "0?dialanika") in new
stack
    -- Executing [000972592603325 at 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 at 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?