It’s been almost a year since we last wrestled with VoIP security for Asterisk®. With Christmas just around the corner, it seemed like a fitting time for a report card. Suffice it to say, the bad guys have not stood still. Attacks have become much more frequent and more sophisticated as VoIP systems have proliferated. A year ago we saw brute force attacks with thousands of password attempts on VoIP servers. These attacks could easily be detected by Fail2Ban. What we are seeing today are one and two hit drive-bys that usually are initiated from Windows zombies or hosted accounts established with stolen credit cards. These VoIP attacks fly under the radar unless you review your logs every day. Have the creeps gotten more patient? No, just smarter. They now understand the VoIP security model that has been deployed on systems like PBX in a Flash, and they simply work around it. Two hits per server, and they’re off to the next IP address only to return in a few hours to try two more. Are these attempts successful? Well, here’s the latest recipient of a $100,000 phone bill so the answer would appear to be affirmative.
We continue to wrestle with new security approaches to better protect Asterisk VoIP systems, and we’ve stumbled upon another golden arrow for your security quiver. Our Incredible PBX platform continues to offer the very best security solution because it is designed to sit safely behind a hardware-based firewall with virtually no exposure to the Internet. But such deployments assume that both your server and your phones are all safely ensconced behind a hardware-based firewall. If it turns out that you want to deploy a SIP phone for use by grandma or you’ve decided you’d like to try hosted PBX service from a provider such as rentpbx.com,1 then there either need to be holes opened in the firewall or there is no hardware firewall protection in the case of hosted service.
Over the past few weeks, we’ve explored a number of new security approaches to better protect your Asterisk server. These include The SunshineNetworks Knock as well as VoIP Black Lists and VoIP White Lists. If you’re technically savvy, you’ll want to carefully consider “The Knock” for all of your SIP phones exposed to the Internet.
We spent a good bit of time considering various VoIP BlackList solutions. As the name implies, a list of the bad guys’ IP addresses is fed into IPtables which then blocks access to your server from these addresses. Sounds good, right? One approach with a BlackList is to block all IP addresses from “problem countries.” The methodology to implement this solution can be found in this thread on the PIAF Forums. The problem, of course, is identifying the “problem countries.” Another option was to implement an IPtables Blacklist based upon the work of the VoIP Blacklist Project. Perhaps ironically, the VoIP Blacklist Project actually blocks the IP addresses of both Nerd Vittles and PBX in a Flash, and emails requesting removal of our IP address were ignored. To save time, the VoIP Blacklist Project employs CIDR Masks which can blacklist hundreds of thousands of IP addresses in one fell swoop. Problem is that a lot of innocent people get caught in the net, and there’s no easy way out without maintaining the blacklist yourself. The final dagger in the black list approach is zombies. Insecure Windows machines have been compromised by the droves worldwide and particularly in the United States. So identifying all of these now-malicious systems is not unlike playing Whack-a-Mole. When you block one of them, six more pop up. So, after giving it the good old college try, our view of VoIP Blacklists should be obvious. No, thanks. There are very real risks that the bad guys can and have poisoned existing blacklists with safe IP addresses, and the number of Windows zombies grows geometrically making it all but impossible to have or maintain a blacklist that affords any real protection.
These results with black lists led us to the conclusion that the only real security mechanism that could protect many VoIP servers today was a VoIP WhiteList for IPtables. As the name implies, we want to identify the IP addresses of every SIP and IAX trunk and extension on your server and then feed those addresses into IPtables so that the only access to VoIP resources on your server is from these addresses. Today’s VoIP WhiteList for IPtables consists of two bash scripts: one queries the MySQL database in which FreePBX stores all of the trunk and extension information for your server and the other populates IPtables with the results of the queries. We would hasten to add that a similar white list is equally important for SSH access to your server although we think it is better to implement an SSH WhiteList on your hardware-based firewall. In this way, you can adjust the SSH white list via web browser while traveling without locking yourself out of your Asterisk server.
Prerequisites. To use today’s VoIP WhiteList for IPtables, you’ll need either a current version of PBX in a Flash or Incredible PBX. Other aggregations will also work provided your system is FreePBX-based (version 2.6 or later), has IPtables already installed and functioning properly, and has an /etc/sysconfig/iptables configuration file that closely matches the stock PBX in a Flash design. We’ll leave it to you to make that call after reviewing the scripts.
VoIP WhiteList Design. We’ve designed the VoIP WhiteList for IPtables to be modular. There’s a firewall-whitelist-gen.sh script which extracts from MySQL the list of IP addresses used by your trunks and extensions. This text-based list is stored in /etc/firewall.whitelist. You can manually add and delete entries from the list once it is populated.You also can rerun the script at any time to generate a fresh catalog of WhiteList IP addresses based upon your current trunk and extension settings. This script also enables access to your server from the public IP address of your server as well as all non-routable IP addresses. Finally, it modifies /etc/sudoers slightly so that Travelin’ Man can be used to add dynamic IP addresses on the fly. We’ll cover that below.
The second script is firewall-whitelist.sh, and it is used to actually implement your new VoIP WhiteList in IPtables. The changes take effect immediately. It also can be run again to update these entries if you manually add or delete IP addresses in /etc/firewall.whitelist. This script always creates a backup copy of your previous /etc/sysconfig/iptables file and names it iptables.timestamp where the timestamp is the date and time of your last update, e.g. iptables.12012010-083841 was created on Dec. 1, 2010 at 08:38:41. If you should ever shoot yourself in the foot, simply copy one of the iptables backup files to /etc/sysconfig/iptables and then restart IPtables: service iptables restart.
WARNINGS: In order to implement the WhiteList, the script removes the existing IPtables entries which permit SIP and IAX access from anywhere using UDP ports 4569 and 5000 to 5082. If you have edited these entries in any way, you’ll need to remove them and restart IPtables before running firewall-whitelist.sh. Otherwise, your more general firewall entries will leave your system vulnerable to access from IP addresses not in your VoIP WhiteList.
If your system is running on a hosted server, you’ll need to make a couple of additions to /etc/sysconfig/iptables and restart IPtables (service iptables restart) before running firewall-whitelist.sh, or you may lock yourself out of your own server. Be sure to add the public IP address of your server, and also add the IP address from which you are making changes to your server. Each entry should look like the following example using your actual IP addresses. And the entries should be added above the COMMIT line in the same section of the iptables file as the existing UDP 10000:20000 ACCEPT entry:
-A INPUT -s 220.127.116.11 -j ACCEPT
Installing the VoIP WhiteList for IPtables. Installation is easy. Just log into your server as root and issue the following commands:
tar zxvf firewall-whitelist.tar.gz
If you installed one of the beta versions of the VoIP WhiteList from the PIAF Forums, then you’ll need to do a little housecleaning before actually running either of the scripts. Just edit /etc/sysconfig/iptables and clean out all of the entries that contain 5000:5082 as well as any entries nearby that include the non-routable IP addresses, e.g. 192.168.0.0. Finally, if there are entries beginning with -A WHITELIST, delete those as well. Then restart IPtables: service iptables restart. Thank you for your testing and feedback!
Deploying Remote SIP Phones. What remains is some method for connecting remote SIP phones with dynamic IP addresses. Our Travelin’ Man application was specifically designed to provide this support although the initial version only opened the necessary IP address for Asterisk access. The latest release also provides the necessary IPtables support. You have two options: either remove the old version and supporting directories under /var/www/travelman or edit the index.php file in each subdirectory you’ve created and make the change shown in this post on the PIAF Forums. Enjoy!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
Some Recent Nerd Vittles Articles of Interest…
- We gratefully acknowledge the contributions of rentpbx.com to the PBX in a Flash Development Team. In addition to hosted accounts to test PBX in a Flash in the hosted environment, rentpbx.com also has contributed technical assistance particularly as it relates to our Google Voice-Asterisk integration efforts. [↩]