James

Thanks for the explaination. It's good to better understand, how to counter 
these attacks.

I need to dig a bit into these asterisk ACL settings to see if it is 
possible to give a range of peer IPs (as the external ones are on dynamic 
IP).

Otherwise, the adaptive ban also seems to work nicely.

Cheers

Michael

James Babiak wrote:

> Michael,
> 
> While obviously you'll want to block these attacks when you see them, as
> long as you use secure credentials for these remote extensions, you
> shouldn't have to worry too much about this attacker actually managing
> to compromise a SIP account. While you'll probably want to keep the
> extension numerical (though you don't need to), the password should be
> secure. That will eliminate 99% of attacks from being successful,
> limiting it just to a nuisance.
> 
> While a dedicated and direct attack (for example: from an employee,
> etc.) against your PBX changes things a bit, almost all attempts are
> random people/groups on the Internet scanning to find any IP that's
> listening on 5060UDP. Once they find you, they will attempt to scan for
> easily compromisable accounts.
> 
> These attackers are going to try very simple brute force attacks. They
> don't know if the extension is valid or not, so they aren't going to
> invest too much time trying to brute force every single possible
> password on every single possible account. Even one account could take
> eons to work through every possible combination of characters. They will
> most likely be trying the extension number for the password, 'password',
> 'secret', etc. In my opinion, the most common error in that record is
> using the extension number for the password (ie: 1234:1234), which
> happens all too frequently.
> 
> You can also use Asterisk's built in peer ACL settings in the sip
> configuration file to prohibit registrations from non-specified IP
> addresses. You should always enable this for all local extensions, and
> if the remote phones will be coming from static IP addresses (or static
> netblocks) you can do that for them as well. If the latter is possible,
> then outside of a security vulnerability in the host system, Asterisk
> itself, or the client location, you can pretty much lock down the pbx
> from remote access attacks.
> 
> And as discussed, you can use the firewall plugins to prevent connection
> attempts like this in the future. Including blacklisting the IPs you see
> trying to probe your system.
> 
> -James
> 
> On 07/20/2010 03:29 AM, Michael wrote:
>> I used the sip-voip plugin. It worked fine. However, security is not
>> enough, it seems to me. I am experiencing hacker attacks on the open port
>> 5060.
>>
>> So, I am wondering, what could be a better solution. Maybe would be
>> interesting to not use port 5060 for external devices. Then the firewall
>> would need to convert it to 5060 for incoming connections. Is this
>> possible?
>>
>> Thanks
>>
>> Michael
>>
>> P.S.:
>> Attacks come from 204.119.22.247, trying dozens of username/password
>> combinations per second. Just a matter of time until they find a valid
>> combination. At the moment, I blocked all external devices (there are
>> only two anyway).
>>
>> Philip Prindeville wrote:
>>
>>    
>>> On 7/11/10 12:13 PM, Lonnie Abelbeck wrote:
>>>      
>>>> On Jul 11, 2010, at 1:04 PM, Philip Prindeville wrote:
>>>>
>>>>
>>>>        
>>>>>> Pass EXT->Local | UDP | Source: 0/0 | Port: 10000-20000
>>>>>>
>>>>>> (The port range here should exactly match your /etc/asterisk/rtp.conf
>>>>>> rtpstart-rtpend port range.  Alternatively you can enable the
>>>>>> 'sip-voip' plugin, but personally I keep the 'sip-voip' plugin
>>>>>> disabled and use the above firewall rule.)
>>>>>>
>>>>>> Hope this helps.
>>>>>>
>>>>>> Lonnie
>>>>>>
>>>>>>
>>>>>>            
>>>>> The problem with this is it opens up ALL ports 10000-20000, not just
>>>>> those that are being used by RTP.
>>>>>
>>>>> I really, really recommend using the SIP-VOIP plugin instead.
>>>>>
>>>>> -Philip
>>>>>
>>>>>          
>>>> In practice I use a *much* smaller port range for RTP, rather than the
>>>> default 10000-20000.
>>>>
>>>> Opening a very small UDP port range for RTP is not a problem for me.
>>>>
>>>> Yes, I know you like the "sip-voip" plugin. :-)
>>>>
>>>> Lonnie
>>>>
>>>>        
>>> What's not to like about it?  :-)
>>>
>>> More to the point, I like exposing only the barest minimal attack
>>> surfaces whenever I can.
>>>
>>>
>>>
>>>
>>>      
>> 
------------------------------------------------------------------------------
>>    
>>> This SF.net email is sponsored by Sprint
>>> What will you do first with EVO, the first 4G phone?
>>> Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first
>>>      
>>
>>
>> 
------------------------------------------------------------------------------
>> This SF.net email is sponsored by Sprint
>> What will you do first with EVO, the first 4G phone?
>> Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first
>> _______________________________________________
>> Astlinux-users mailing list
>> Astlinux-users@lists.sourceforge.net
>> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>>
>> Donations to support AstLinux are graciously accepted via PayPal to
>> pay...@krisk.org.
>>    
> 
> 
------------------------------------------------------------------------------
> This SF.net email is sponsored by Sprint
> What will you do first with EVO, the first 4G phone?
> Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first



------------------------------------------------------------------------------
This SF.net email is sponsored by Sprint
What will you do first with EVO, the first 4G phone?
Visit sprint.com/first -- http://p.sf.net/sfu/sprint-com-first
_______________________________________________
Astlinux-users mailing list
Astlinux-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/astlinux-users

Donations to support AstLinux are graciously accepted via PayPal to 
pay...@krisk.org.

Reply via email to