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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