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.
