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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>Actually even the Security log (and AMI security event) is nothing more than failed dial/register attempts against Asterisk. There is no awareness of corrupt SIP attacks, detection of polling for insecure extensions, goefencing based on source IP (why allow connections from Russia if all of your uses are in Texas), detection of rapid dialing rates once connected to an IVR, etc.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>So your entire security system is based on Asterisk saying a dial/register failed. That’s a small fraction of the attack types against, and attack surface offered by, PJSIP/SIP/Asterisk. Even worse, if you run a configuration generator (eg FreePBX)..well…do a google search to see the exploits that are published regularly. I realize FreePBX/Sangoma now owns Digium so that discussion should probably go no further.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>So don’t get me wrong….fail2ban is way better than nothing. But it may instill a false sense of security. And that was Digium’s point in the post. So if the OP needs a free and fast solution against simple script kiddie attacks then installing fail2ban is a big thumbs up in my opinion.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>There have been similar discussions in other groups as to why even have a firewall, since you can close ports not needed by your services. There are some people who are very passionate about their view that firewalls are a waste of time and money. Far be it from me to say they’re wrong…but I’ve tried to point them to some interesting articles.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>If you are a pure open source advocate there are still a lot more tools you can use to secure you PBX. Think SNORT, I think pfsense offers a free database that’s accurate to a country level, etc. If you want commercial then there are even more options. But that’s the wrong forum for the biz stuff<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>I feel I tread on the edge of a holy war </span><span style='font-size:11.0pt;font-family:Wingdings;color:#1F497D'>J</span><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'> So I’ll leave my thoughts here and go no further<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'> asterisk-users [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Matthew Jordan<br><b>Sent:</b> Wednesday, August 29, 2018 10:52 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject:</b> Re: [asterisk-users] getting invites to rtp ports ??<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group <<a href="mailto:support@telium.ca">support@telium.ca</a>> wrote:<o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'><p class=MsoNormal style='margin-bottom:12.0pt'>Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened numerous time). If you are comfortable hacking chan_sip.c you may prefer to get the same messages from the AMI. It still misses a lot but that approach is better than nothing.<br><br>Digium warns not to use fail2ban / log trolling as a security system: <a href="http://forums.asterisk.org/viewtopic.php?p=159984" target="_blank">http://forums.asterisk.org/viewtopic.php?p=159984</a><br><br><o:p></o:p></p></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>That's some pretty old advice.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>The rationale for *not* using general log messages with fail2ban still stands: the general WARNING/NOTICE/etc. log messages are subject to change between versions, and no one wants that to impact someone's security. So you should not use those messages as input into fail2ban.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>That rationale did lead to the 'security' event type in log messages. Security Event Logging - as it is called - got added into Asterisk quite some time ago. So long ago I'm really not sure which version. At a minimum, Asterisk 11, but I'm pretty sure it was in 10 as well.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Documentation for it can be found here:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger">https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>And here:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><a href="https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration">https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration</a><o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Note that this also fires off AMI events (and ARI events, IIRC).<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>If, for whatever reason, you do not get a SECURITY log message or a corresponding event when something 'bad' happens, that would be worth some additional discussion. If anything, the events can be a bit chatty...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt'><p class=MsoNormal>-----Original Message-----<br>From: asterisk-users [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of sean darcy<br>Sent: Wednesday, August 29, 2018 6:33 PM<br>To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] getting invites to rtp ports ??<br><br>On 08/29/2018 11:59 AM, Telium Support Group wrote:<br>> Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki:<br>> <br>> <a href="https://www.voip-info.org/asterisk-security/" target="_blank">https://www.voip-info.org/asterisk-security/</a><br>> <br>> <br>> <br>> -----Original Message-----<br>> From: asterisk-users [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <br>> On Behalf Of sean darcy<br>> Sent: Wednesday, August 29, 2018 10:46 AM<br>> To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>> Subject: Re: [asterisk-users] getting invites to rtp ports ??<br>> <br>> On 08/29/2018 09:42 AM, Carlos Rojas wrote:<br>>> Hi<br>>><br>>> Probably somebody is trying to hack your system, you should block <br>>> that ip on your firewall.<br>>><br>>> Regards<br>>><br>>> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <<a href="mailto:seandarcy2@gmail.com" target="_blank">seandarcy2@gmail.com</a> <br>>> <mailto:<a href="mailto:seandarcy2@gmail.com" target="_blank">seandarcy2@gmail.com</a>>> wrote:<br>>><br>>> I'm getting invites to very high ports every 30 seconds from a<br>>> particular ip address:<br>>><br>>> Retransmitting #10 (NAT) to <a href="http://5.199.133.128:52734" target="_blank">5.199.133.128:52734</a><br>>> <<a href="http://5.199.133.128:52734" target="_blank">http://5.199.133.128:52734</a>>:<br>>> SIP/2.0 401 Unauthorized<br>>> Via: SIP/2.0/UDP<br>>> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734<br>>> From: <<a href="mailto:sip%3A37120116780191250@67.80.191.250" target="_blank">sip:37120116780191250@67.80.191.250</a><br>>> <mailto:<a href="mailto:sip%253A37120116780191250@67.80.191.250" target="_blank">sip%3A37120116780191250@67.80.191.250</a>>>;tag=1872048972<br>>> To: <<a href="mailto:sip%3A3712011972592181418@67.80.191.250" target="_blank">sip:3712011972592181418@67.80.191.250</a><br>>> <mailto:<a href="mailto:sip%253A3712011972592181418@67.80.191.250" target="_blank">sip%3A3712011972592181418@67.80.191.250</a>>>;tag=as3a52e748<br>>> Call-ID: 1504207870-295758084-609228182<br>>> CSeq: 1 INVITE<br>>> .......<br>>> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on<br>>> 1504207870-295758084-609228182...<br>>><br>>> I thought invites had to go to port 5060 or so. I don't understand<br>>> why somebody (let's assume a bad guy) is trying ports above 50000.<br>>><br>>> sean<br>>><br>>><br>> <br>> Ok, so the high port is not the destination port but the source port.<br>> <br>> So I hacked the log warning in chan_sip.c on non-critical invites to show the source ip:<br>> <br>> ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from <br>> %s.\n",<br>> pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));<br>> <br>> With that in the log, I'm now blocking the ip addresses.<br>> <br>> Thanks,<br>> sean<br>> <br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>> <br>> Astricon is coming up October 9-11! Signup is available at: <br>> <a href="https://www.asterisk.org/community/astricon-user-conference" target="_blank">https://www.asterisk.org/community/astricon-user-conference</a><br>> <br>> Check out the new Asterisk community forum at: <br>> <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br>> <br><br>I agree. That's why I hacked chan_sip.c to get the addresses in the log.<br><br>I'm surprised they're not in the log by default. I must be the only person who gets these "non-critical invites".<br><br>sean<br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Astricon is coming up October 9-11! Signup is available at: <a href="https://www.asterisk.org/community/astricon-user-conference" target="_blank">https://www.asterisk.org/community/astricon-user-conference</a><br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Astricon is coming up October 9-11! Signup is available at: <a href="https://www.asterisk.org/community/astricon-user-conference" target="_blank">https://www.asterisk.org/community/astricon-user-conference</a><br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></blockquote></div><p class=MsoNormal><br clear=all><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <o:p></o:p></p><div><p class=MsoNormal>Matthew Jordan<br>Digium, Inc. | CTO<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><o:p></o:p></p></div></div></div></body></html>