Thanks again. I really appreciate any advice that can help me identify how they gained access. I don't think it's a process of them gaining access through my DID. I enabled the full logging in logger.conf and a few things popped up in the full log which show me a few things. I have limited knowledge about all this so I could use more input on what this means. But apparently they did something to gain access by trying to register several IP address at once.
Aug 29 23:11:51] NOTICE[92568] chan_sip.c: Registration from '"94.23.222.75:5060.....85.31.178.110.....203.174.41.18....190.10.27.80"<sip:[email protected]>' failed for '188.161.221.100' - No matching peer found [Aug 30 00:37:40] NOTICE[92568] chan_sip.c: Registration from '85.43.196.74 ... 87.236.186.110...202.43.190.195..202.43.190.195..203.215.155.38<sip:[email protected]>' failed for '109.253.85.228' - No matching peer found [Aug 30 00:37:40] NOTICE[92568] chan_sip.c: Registration from '85.43.196.74 ... 87.236.186.110...202.43.190.195..202.43.190.195..203.215.155.38<sip:[email protected]>' failed for '109.253.85.228' - No matching peer found NOTICE HERE THE LOGIN FOR EXT #100 FAILS BECUASE THERE IS NO EXT #100 BUT ONLY 5 SECONDS LATER THEY WERE IN AND DIALING A CALL IP ADDRESS 109.253.85.228 ORIGINATES IN ISRAEL [Aug 30 00:37:55] VERBOSE[92568] logger.c: -- Executing [011972599544...@default:1] Set("SIP/98.242.233.74-00000004", "CALLERID(all)=xxxxxxxxxxx") in new stack [Aug 30 00:37:55] VERBOSE[92568] logger.c: -- Executing [011972599544...@default:2] Dial("SIP/98.242.233.74-00000004", "SIP/xxx/011972599544327,,wWFotThH") in new stack [Aug 30 00:37:55] VERBOSE[92568] logger.c: -- Called xxx/011972599544327 [Aug 30 00:37:56] VERBOSE[92568] logger.c: -- SIP/xxx-00000005 is making progress passing it to SIP/98.242.233.74-00000004 [Aug 30 00:37:58] VERBOSE[92568] logger.c: -- Got SIP response 402 "Zero balance" back from 204.74.213.5 [Aug 30 00:37:58] VERBOSE[92568] logger.c: -- No one is available to answer at this time (1:0/0/0) [Aug 30 00:37:58] VERBOSE[92568] logger.c: -- Executing [011972599544...@default:3] PlayTones("SIP/98.242.233.74-00000004", "congestion") in new stack [Aug 30 00:37:58] VERBOSE[92568] logger.c: -- Executing [011972599544...@default:4] Hangup("SIP/98.242.233.74-00000004", "") in new stack [Aug 30 00:37:58] VERBOSE[92568] logger.c: == Spawn extension (default, 011972599544327, 4) exited non-zero on 'SIP/98.242.233.74-00000004' [Aug 30 00:38:00] NOTICE[92568] chan_sip.c: Registration from '85.43.196.74 ... 87.236.186.110...202.43.190.195..202.43.190.195..203.215.155.38<sip:[email protected]>' failed for '109.253.85.228' - No matching peer found ________________________________ From: Vahan Yerkanian <[email protected]> To: Asterisk on BSD discussion <[email protected]> Sent: Mon, August 30, 2010 9:42:35 AM Subject: Re: [Asterisk-bsd] Securing Asterisk with a DID On 8/30/10 4:34 PM, Frank Griffith wrote: > Executing [011972599544...@default:1] This is perhaps one of the worst things you can ever do with Asterisk - putting toll access into the default context. Never put anything you don't want to be accessible to unauthenticated guests there. Your Asterisk server with that config is an open gateway, and anyone can dial through it if they try to dial SIP/011someth...@your_ip. Solution: move everything out of the default context in extensions.conf or .ael, leaving it empty, and place all the extensions instead in a different context. Assign your devices and/or DID accounts to that context so the extensions are still available to them, f.e. [myDIDprovider] type=user host=ipaddr_or_hostname context=my_context disallow=all allow=whatever_codec(s) qualify=yes [201] ; a sip account type=friend host=dynamic secret=verysecretandlonghash context=my_context disallow=all allow=whatever_codec(s) qualify=yes These are rough examples, but should be enough for the start. Yeah, and make sure you have alwaysauthreject=yes in sip.conf Hope this helps, Vahan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Asterisk-BSD mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-bsd
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