One of the things I also do is block out all international numbers except the ones I need... typically in my business (hotel net / telecom) we see VOIP systems hacked for use by drug deals, terrorist activity etc.. many calls are made to somalia, afghanistan, colombia etc.. so typically in all of our systems we kill out those countries and several others that are known hot spots for hackers to make their destination calls to....
If you don't call but just a few internal countries (or none) - kill them in your dialplan, asterisk wont dial a number it cannot match regardless of whether they get access to one of your SIP station accounts or not... Another asterisk attack that is common is that many people use default configs that have the asterisk manager interface open and if that is hacked, then crackers can add rules to the dialplan on the fly and open your asterisk system up... make sure those ports (typically 5038) is locked down in your firewall... -Christopher -----Original Message----- From: Cleve Jansen [mailto:clev...@gmail.com] Sent: Wednesday, October 13, 2010 7:45 PM To: 'AstLinux Users Mailing List' Subject: Re: [Astlinux-users] Firewall Question Dan, What you are seeing is sip hacking where they have run a scan on your ip and found port 5060 open, so they are trying to attempt to register a sip device on your setup so they could make very expensive calls. I know Astlinux does not support this but in my normal installs I use fail2ban plus CSF (Config system firewall) which keeps this sort of thing away as it on the rise, if you do a search on google for "sip hacking" you will find some interesting information. What I would do in your case as you have a remote extension, is this extension on a static ip if so you could add the following to that extension. Example deny=0.0.0.0/0.0.0.0 permit=your.remote.static.ip/255.255.255.0 Additionally in sip.conf you can add the following too, sometimes if the remote extension is not on a static ip I use this. alwaysauthreject=yes allowguest=no The above is a few things I implement and also a few others where I cannot add fail2ban or CSF. Hope this helps.. Good Luck!! Cleve -----Original Message----- From: Dan Ryson [mailto:d...@ryson.org] Sent: Thursday, 14 October 2010 9:02 AM To: astlinux-users@lists.sourceforge.net Subject: Re: [Astlinux-users] Firewall Question On 10/13/2010 3:34 PM, Philip Prindeville wrote: > On 10/13/10 7:44 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. > That would be redundant. > > There's already a generic way to limit UDP traffic to the firewall. Adding a second method to do the same would create ambiguity and confusion. > > All the sip plugin does--all it's supposed to do--is maintain a NAT hole for the SDP (media) stream open based on the information it sees in the SIP transactions (since the SDP endpoints talk to each other, but the SIP stream goes through one or more intermediaries). > > Given the asymmetry of SIP to SDP, you can't have SDP maintain its own associations in NAT, especially not if you use short timers and VAD or muting. > > The SIP plugin is a hook for a NAT helper. Period. It doesn't do access control because it doesn't need to. > > Dan: I'd try using $HOST_OPEN_UDP. The interface is your $EXTIF. The source-ip is whatever source address you want, the destination port is 5060. > > Also, in sip-voip.conf, set: > > SIP_VOIP_PORTS="5060" > > You only need addition ports if your Asterisk is listening on alternate ports, which it probably isn't. > > Lonnie: I might have stepped out of Astlinux, but not from AIF. Please don't be modifying my plugins without my consent. > > -Philip Hello Philip, Thank you for the explanation regarding the sip plugin. Just for some background, the problem I'm trying to solve relates to uninvited SIP traffic that I'm having difficulty explaining. Adaptive-ban kills off any uninvited guests in a short time but I would prefer the firewall to be my first line of defense - and have adaptive-ban serve as a safety-net. Perhaps that's not the best approach. I hope you (and anyone else who may wish to chime in) won't hesitate to correct me if you disagree. At present, only a single IP phone, which is on a fixed IP, needs access to Asterisk, so I've opened only port 5060 for that distant, fixed address. I'm also using a route-able IP address on the AstLinux WAN port so I'm not sure that I'll need any help with NAT. Since much of this is over my head, I'm not certain how to troubleshoot this apparent problem. Therefore, I'm thankful that you and Lonnie have both volunteered to help me solve this problem. Please consider that there's a good chance I've fat-fingered something and caused the troubles myself. With kind regards, Dan ---------------------------------------------------------------------------- -- Beautiful is writing same markup. Internet Explorer 9 supports standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. 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 _______________________________________________ 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. ---------------------------------------------------------------------------- -- Beautiful is writing same markup. Internet Explorer 9 supports standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. 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 _______________________________________________ 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. ------------------------------------------------------------------------------ Beautiful is writing same markup. Internet Explorer 9 supports standards for HTML5, CSS3, SVG 1.1, ECMAScript5, and DOM L2 & L3. 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 _______________________________________________ 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.
