That's fantastic, Lonnie.

Thank you for bringing this up with AIF.  Hopefully, this will come to 
fruition!

Dan

On 10/13/2010 1:33 PM, Lonnie Abelbeck wrote:
> Dan,
>
> A new config variable, SIP_VOIP_REMOTE_HOSTS has been added to the sip-voip 
> plugin in the next AIF.
>
> https://rocky.eld.leidenuniv.nl/trac/aif/changeset/434/
>
> Thanks for the suggestion.
>
> Lonnie
>
>
> On Oct 13, 2010, at 10:26 AM, Dan Ryson wrote:
>
>>   So it's that simple?  I really like simple.
>>
>> Adaptive-ban has been very effective.  However, since I only have the
>> one outside user, I'd also like to block the ports at the firewall.
>>
>> Thanks as always for your insight.
>>
>> Dan
>>
>> On 10/13/2010 10:44 AM, Lonnie Abelbeck wrote:
>>> On Oct 13, 2010, at 9:15 AM, Dan Ryson wrote:
>>>
>>>>   All,
>>>>
>>>> I wonder if I may, once again, ask for your help.
>>>>
>>>> Using the GUI to configure the firewall, my intent was to open only one
>>>> "Source IP" to port 5060, for an off-site IP phone.  I'm depending on
>>>> frequent&   regular registration traffic to keep port 5060 open to
>>>> providers.  Despite this, I see the occasional registration attempt from
>>>> elsewhere, as shown below.
>>>>
>>>> Oct 13 04:23:36 sip local0.notice asterisk[2776]: NOTICE[2776]: 
>>>> chan_sip.c:16474 in handle_request_register: Registration from 
>>>> '"1010161682"<sip:1010161...@169.25.161.29>' failed for '140.117.176.226' 
>>>> - No matching peer found
>>>>
>>>>
>>>> So, with all other source IPs closed to port 5060, how might a
>>>> registration request from '140.117.176.226' be reaching Asterisk?
>>>>
>>>> The only thing that looked a bit suspicious in iptables, is this:
>>>>
>>>> Chain EXT_INPUT_CHAIN (2 references)
>>>> target     prot opt source               destination
>>>> ACCEPT     udp  --  anywhere             anywhere            udp 
>>>> dpts:5060:5080
>>>>
>>>>
>>>> However, it looks like the above is merely the result of settings in the
>>>> SIP-VOIP plugin, which specifies ports 5060:5080.  When disabling
>>>> SIP-VOIP, the above entry goes away.
>>>>
>>>> Your thoughts?
>>>>
>>>> Thanks for considering my question.
>>>>
>>>> Dan
>>> Don't enable the sip-voip plugin. :-)
>>>
>>> The sip-voip plugin may have it's place, (it basically automatically opens 
>>> the RTP voice ports) but I personally don't enable it.
>>>
>>> So, if you disable the sip-voip plugin you will need to allow a UDP range 
>>> matching your asterisk rtp.conf range. (make it smaller than the default)
>>>
>>> Or, keep the sip-voip plugin enabled and also enable the adaptive-ban 
>>> plugin to ban the attack probes.
>>>
>>> Lonnie
>>>
>>> PS: A better long term solution would be to add a SIP_VOIP_SOURCE="0/0" 
>>> variable to the sip-voip plugin, so you can limit by the source address... 
>>> I'll try to get that in the next version of AIF.
>>>
>>>
>>>
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Spend less time writing and  rewriting code and more time creating great
experiences on the web. Be a part of the beta today.
http://p.sf.net/sfu/beautyoftheweb
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