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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> 
> ------------------------------------------------------------------------------
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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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