For such cases i created a dialplan in the default dialplan which blocks the ip of the hacker with iptables.
On Monday, June 8, 2015, Luca Bertoncello <lucab...@lucabert.de> wrote: > 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 <javascript:;>) > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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