[Asterisk-bsd] Securing Asterisk with a DID

Mon Aug 30 13:13:30 CDT 2010

Well obviously there is a remote SIP connecting. But my server is not setup to 
allow any remote connections. According to the VOIP provider I've been brute 
force attacked yet Asterisk leaves no log information as to which account was 
logged into. Thus I'm still stuck trying to figure out what happened.

My server does login to my VOIP provider via sip. So if I understand your 
question correctly the answer is no. I do not have nor do I allow any remote SIP 
connections, yet the evidence is there that someone connected remotely and 
dialed some numbers.

I guess this is showing how security is still a very misunderstood topic.

________________________________
From: Giancarlo Rubio <gianrubio at gmail.com>
To: Asterisk on BSD discussion <asterisk-bsd at lists.digium.com>
Sent: Mon, August 30, 2010 10:24:22 AM
Subject: Re: [Asterisk-bsd] Securing Asterisk with a DID

Did you have remote sip that connect in your asterisk, not your asterisk to 
remote?? 

If no drop port 5060 in your external interface via firewall rule like

block drop on $ext_if from any to $ext_ip port 5060

2010/8/30 Frank Griffith <glassdude45 at yahoo.com>

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:100 at 98.242.233.74>'
> 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:100 at 98.242.233.74>'
> 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:100 at 98.242.233.74>'
> 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 
>[011972599544327 at 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 
>[011972599544327 at 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 
>[011972599544327 at default:3] PlayTones("SIP/98.242.233.74-00000004", 
>"congestion") in new stack
>[Aug 30 00:37:58] VERBOSE[92568] logger.c:     -- Executing 
>[011972599544327 at 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:100 at 98.242.233.74>'
> failed for '109.253.85.228' - No matching peer found
>
>
>
>
________________________________
From: Vahan Yerkanian <vahan at arminco.com>
>To: Asterisk on BSD discussion <asterisk-bsd at lists.digium.com>
>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 [011972599544327 at 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/011something at 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
>
>
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-- 
Giancarlo Rubio

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