Posts tagged: security

Avoiding a $100,000 Phone Bill: VoIP WhiteList for IPtables

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 222.222.222.222 -j ACCEPT

Installing the VoIP WhiteList for IPtables. Installation is easy. Just log into your server as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/firewall-whitelist.tar.gz
tar zxvf firewall-whitelist.tar.gz
./firewall-whitelist-gen.sh
./firewall-whitelist.sh

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…

  1. 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. []

The Incredible PBX: Meet the New Kid on the Block

As much as we loved the moniker, the Orgasmatron build was in desperate need of a name change to more accurately describe its true heritage. We didn't look too far for just the right name. Meet The Incredible PBX!

Thanks to the Zero Internet Footprint™ design, it's the most secure Asterisk®-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server.1 And you won't find a more full-featured Personal Branch Exchange™.

NEWS FLASH: The Incredible PBX is now available for Asterisk 1.8! Go here.

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

The Incredible PBX is much more than just a name change. In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features tailored to meet the needs of the individual: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we'll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

The Incredible PBX Inventory. For those wondering what's included with The Incredible PBX, here's a feature list of components you get in addition to the base install of PBX in a Flash with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using provided scripts.

Prerequisites. Here's what you'll need to get started:

  • Broadband Internet connection
  • $200 PC3 on which to run The Incredible PBX or a Proxmox VM
  • dLink Router/Firewall. Low Cost: $35 WBR-2310  Best: DGL-4500
  • Free Google Voice account (Available in U.S. without an invite at this link)
  • Free SIPgateOne residential account (U.S. cell to get SMS invite) OR
  • Free IPkall IAX account (recommended for international users)

Installing The Incredible PBX. The installation process is simple and straight-forward. Just don't skip any steps. Here are the 5 Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls:

1. Install the latest version of PBX in a Flash
2. Download & run The Incredible PBX installer
3. Set up your two provider accounts
4. Configure a softphone or SIP telephone
5. Run the configure-gv credentials installer

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest 32-bit version of PBX in a Flash. This new build works much better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.5 operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use virtually identical payloads for all versions of PBX in a Flash.

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Or, if you are feeling really adventurous or if you have new, bleeding edge hardware, try our new 32-bit, PIAF 1.7 build which features CentOS 5.5. This new release is available from SourceForge or Google Docs. The MD5 checksum for the PIAF 1.7 build is 184cdb00142ccdd814b11de23fb00082.

Download the brand-new 32-bit PIAF 1.7.5.5. from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and type ksalt press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can't find the kickstart file. Just tab to OK and press Enter. Don't change the name or location of the kickstart file! This will get you going. Think of it as a CentOS 'feature'. :-)

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A choose PIAF-Silver option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes
status

When prompted, change the ARI password to something really obscure. You're never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time. Write down the dynamic IP address assigned to your server after running the status command. You'll need it shortly.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source with The Incredible PBX. The version of Asterisk installed from our payload file is very stable.

Running The Incredible PBX Installer. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx.x
chmod +x incrediblepbx.x
./incrediblepbx.x

Have another 15-minute cup of coffee. It's a great time to consider a modest donation to the Nerd Vittles project. You'll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Here's a short video demonstration of the Incredible PBX installer process:

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. If you are eligible, we strongly recommend a SIPgate One residential account for The Incredible PBX. However, you may elect to use an IPkall account as an alternative. Both are free; however, you cannot register The Incredible PBX to IPkall's servers so you'll need to punch a hole in your firewall to receive incoming calls from Google Voice and IPkall. This step is not necessary with SIPgate accounts since there is a permanent registered connection between The Incredible PBX and SIPgate's servers!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we'd recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You'll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don't worry. You can erase your cellphone number from your account once it is set up and working properly. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn't matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you'll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You'll need these in a few minutes to complete the configuration of The Incredible PBX. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you're using IPkall as your intermediate provider, first log in to your hardware-based firewall/router and map UDP port 45694 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don't worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you're using SIPgate instead of IPkall.

After your firewall is properly configured, you'll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we'll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don't have an available number, your request apparently goes in the bit bucket. You'll know because IPkall typically turns these requests around in a few minutes. Don't worry about the mothership entry. We'll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it's dynamic, then you'll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you've got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you'll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It's worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it's a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Configuring Google Voice. Google Voice no longer is by invitation only so, if you're in the U.S. or have a friend that is, head over to the Google Voice site and register. After you've chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type.

Google places a test call to your number so you'll have to delay it a bit for IPkall. If you're using SIPgate, go ahead and tell Google to place the test call which will be forwarded to your cellphone. Enter the two-digit code that's displayed when you're prompted to do so. With IPkall, wait until we finish running the credentials configurator below.

While you're still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

  • Call Screening - OFF
  • Call Presentation - OFF
  • Caller ID (In) - Display Caller's Number
  • Caller ID (Out) - Don't Change Anything
  • Do Not Disturb - OFF

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

If you're using SIPgate and you've confirmed your number, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to The Incredible PBX. We're ready to insert your credentials and SIPgate/IPkall information into The Incredible PBX. You'll need several pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 10-digit SIPgate or IPkall RingBack DID. You'll also need to reenter your passwd-master password which is used to configure CallerID Superfecta. Finally, you'll need to tell the configurator whether you're using a SIPgate or IPkall account. In the case of SIPgate, you'll also be prompted to enter your SIP ID and SIP password. These are NOT the same as your account credentials!!

Log back into your server as root and issue the following command to kick off the configurator: ./configure-gv.x. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!! Once you've checked and double-checked your entries, press Enter and The Incredible PBX setup will be completed. You'll need to press Enter again when the script finishes to reboot your PBX. After the reboot, your system will have randomly-generated passwords for every extension and voicemail box that is preconfigured on your system. The DISA password also has been changed. We generate five-digit passwords. If you will sleep better with longer passwords, be our guest. They are easily reset using the FreePBX web interface described elsewhere in this article.

Finally, log back into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone, and you'll find lots of recommendations on Nerd Vittles. For today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

If you're using SIPgate as your provider with Google Voice, you're ready to place a test call. If you're using IPkall, we still need to verify your IPkall number with Google Voice. Return to Google Voice and tell it to place the test call to your IPkall number which you've already entered as your destination number. Your softphone will ring momentarily. Enter the two-digit code provided by Google Voice, and you're all set.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let's place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password, and we're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider. For this week, we'll point you to some things to play with on your new server. Then, in the subsequent articles below, we'll cover in detail how to customize every application that's been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we'd recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there's really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. Be safe!

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PBX in a Flash Support Conference Bridge
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)



Click above. Enter your name and phone number. Press Connect to begin the call.


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and The Incredible PBX adds a bunch of additional options. See if you can find all of them. Also check out Tweet2Dial which uses Twitter to make Google Voice calls, send free SMS messages, and manage your Incredible PBX.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You'll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you've heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud as well. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It's perfect for off-site backups which we'll cover in a few weeks.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Originally published: Monday, April 19, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Adding Skype to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won't have to wait long for an answer to your questions.

Coming Soon. We haven't forgotten. We'll cover setting up multiple Google Voice accounts for simultaneous calling on multiple channels very soon. And the new (free) Skype Gateway to Asterisk for The Incredible PBX is now available. The FreePBX components already are in place to support inbound and outbound calling via Skype. You can even try a test call to our Aspire One Revo today by dialing nerdvittles from your favorite Skype client. Beginning today, this article will be available on http://IncrediblePBX.com. Then Nerd Vittles will return to our (almost) weekly schedule of new articles. 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...

  1. Requires a SIPgate One account. []
  2. For Asterisk 1.6 or for 64-bit systems with Asterisk 1.4 or 1.6, use the Cepstral install procedures outlined in this Nerd Vittles article. []
  3. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. []
  4. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. []

The Incredible PBX: Adding Remotes, Preserving Security

Unlike most Asterisk®-based PBXs which are insecure as installed and leave it to you to implement sufficient safeguards to preserve the integrity of your system, the Incredible PBX is delivered with rock-solid, air-tight security already in place. Because it is designed to operate behind a hardware- based firewall, what you'll be doing when you want to add functionality with the Incredible PBX is loosening security rather than tightening it. The trick, of course, is to do it in a way that doesn't compromise the overall integrity of your system. As delivered, the Incredible PBX relies upon four layers of network security: a hardware-based firewall of your choice1, a preconfigured IPtables software-based Linux firewall, preconfigured Fail2Ban to monitor your logs for suspicious activity and to block specific IP addresses when abuse is detected, and random passwords for all extensions and DISA connections.

If you installed the Incredible PBX using SIPgate as the intermediate provider with Google Voice, then your hardware-based firewall should have no ports opened and forwarded to your server. If you used IPkall, then only UDP 4569 has been opened and forwarded to your server. And the Incredible PBX IPtables setup for IAX restricts access to just a few IP addresses to support IPkall.

There are obviously situations in which you will want or need additional connectivity. The most likely one involves activation of SIP telephones at remote locations, such as a branch office, or Grandma's house or a relative in college. The other obvious use is with cellphones and PDAs that support SIP clients such as Android phones, iPhones, and iPads.2

What we'd recommend you not do is open the SIP floodgate to your PBX by providing unrestricted inbound SIP access, but we'll show you how if you really want or need this functionality. As desirable as this can be, it is accompanied by an array of security issues that really are not worth the risks unless you know what you're doing and you're willing to stay on top of security updates and keep your system patched.

Let's first tackle how to provide limited inbound SIP functionality without selling the farm. If the remote site has a fixed IP address, the procedure to allow remote access to your server is fairly straight-forward: just map the SIP ports on the hardware-based firewall to your server (UDP 5000:5082 and UDP 10000:20000) and then restrict SIP access using IPtables to the remote IP address as well as the subnet of your private LAN. You can decipher your private subnet by running status. If your server's IP address is 192.168.0.123, then your private subnet would be 192.168.0.0. The IPtables firewall settings are stored in /etc/sysconfig/iptables. Edit that file and find the line that looks like this:

-A INPUT -p udp -m udp --dport 5000:5082 -j ACCEPT

Delete or comment out this entry with a leading # and insert new entries that look like the following using the public IP address(es) you wish to add plus the private subnet:

-A INPUT -p udp -m udp -s 141.146.20.10 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 141.146.20.11 --dport 5000:5082 -j ACCEPT
-A INPUT -p udp -m udp -s 192.168.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT


After making the changes, save the file: Ctrl-X, Y, then Enter. Then restart IPtables: service iptables restart.

Unfortunately, in many situations, the remote phone or cellphone uses an Internet connection with a dynamic IP address. So we don't know the actual IP address that will be assigned. There are a number of solutions to this problem, and we'll rank them in our order of preference. First, spend the $200 and install another Incredible PBX at the remote site. Then the two servers can be linked with IAX connections between the servers making connectivity between the systems totally transparent. Second, install VPN routers at both sites and use a private IP address to establish connectivity with the host system. In this situation, you will have the equivalent of a fixed IP address for the remote device which makes it the equivalent of the fixed IP address solution above. Third, install OpenVPN on your host system and purchase a SIP phone or cellphone that supports VPN connectivity. Most of the high-end SNOM SIP phones have this functionality as do Android phones, iPhones, and iPads. With this setup you also have the equivalent of a fixed IP address, even though it's on a virtual private network. Fourth, talk to the Internet service provider at your remote site and obtain the range of IP addresses that DHCP hands out to those using their services... or just make an educated guess.3

BEFORE Activating Full SIP Connectivity. OK. We hear you. You travel for a living, and the IP address of your cellphone changes hourly, all day, every day of the year. Then, yes, you are a candidate for a full-fledged Asterisk server with unlimited SIP access. Before covering how, let's review what responsibilities go with running such a server. Bear in mind that one compromised SIP password or otherwise vulnerable application on your server (including Asterisk, FreePBX, SSH, and hundreds of others), and you may very well be the proud owner of a whopping phone bill. And we're not talking hundreds of dollars. It could very well be tens of thousands of dollars. And it doesn't take weeks or months. It could be a few hours.

Baker's Dozen SIP Security Checklist

1. Keep Asterisk Current & Patched
2. Keep FreePBX Current & Patched
3. Make Frequent Backups
4. Visit PBX in a Flash Forums Regularly
5. Subscribe to PBX in a Flash RSS Feed
6. Secure Alphanumeric Extension Passwords
7. Secure DISA, VMail, Root, FreePBX Passwords
8. Lock Down Extensions with Deny/Permit
9. Turn Off Recurring Payments with Providers
10. Restrict Trunks to 1-2 Simultaneous Calls
11. Tighten Dialplan by Removing Wildcards
12. Eliminate Intl & Toll Calls With Providers
13. Check FreePBX Call Logs Daily for Abuse

Baker's Dozen SIP Security Checklist. Before opening the floodgates, let's review what you need to do. First, you'll need to run the very latest version of Asterisk... all the time. This means you need to monitor asterisk.org, and keep your system up to date by running update-scripts, update-source, and update-fixes regularly. The default version of Asterisk on current PBX in a Flash and Incredible PBX builds is extremely reliable, but it contains SIP and IAX vulnerabilities which should not be exposed directly to the Internet! Second, you need to run the latest version of FreePBX and apply all patches as they are released. Third, you need to make frequent backups appreciating that sometimes the Asterisk and FreePBX developers get things horribly wrong, and stuff that used to work no longer does. Believe it or not, they're human! Fourth, you need to visit the PBX in a Flash Forums daily and keep abreast of security alerts and bug reports on CentOS, Asterisk, and FreePBX. Fifth, you need to subscribe to the PBX in a Flash RSS Feed which provides regular security alerts when there are reported problems. Sixth, you need to really secure your extension passwords with very long, complex alphanumeric passwords. Ditto for your root and FreePBX passwords! Seventh, for DISA and voicemail, these passwords need to be numeric, complex, and extra long. Eighth, you need to lock down as many of your extensions as possible with deny/permit settings to restrict the IP addresses of those extensions. If you only have one or two remote SIP extensions with dynamic IP addresses, then all of the rest should have deny/permit entries! Ninth, turn off recurring payments with all of your telephony providers and keep minimal funds available in all of your accounts. This means you'll have to monitor these accounts to make sure they are not deactivated for lack of funds. Tenth, restrict all of your trunks to one or at most two simultaneous calls to reduce your call exposure in the event someone breaks into your system. Eleventh, tighten up your Trunk Dial Rules and eliminate any entries that would permit calls to anywhere in the world! If you don't regularly make international calls, there's absolutely no reason to have such entries in your dialplan. If you still have Ma Bell PSTN lines, this is even more important. In fact, consider eliminating long distance access to all of these trunks. Twelfth, where possible, configure your provider accounts to eliminate international and toll calls of all varieties. Finally, check your FreePBX call log every day to make certain no one is making calls on your nickel.

If you are unwilling or unable to perform these Baker's Dozen steps while continuing to monitor the sites provided and recheck your setup regularly (at least every week), don't activate unrestricted SIP access to your server.

Other Options. Consider using an intermediate provider such as voip.ms to provide SIP URI access to your server. Keep in mind that having a registered connection between your server and a VoIP provider alleviates the need to punch a hole in your firewall. So the idea here is to sign up for an inexpensive voip.ms account and set up the trunk connection with your server as either an IAX or SIP account with an always-on connection. Then voip.ms gives you the option of activating a SIP URI as part of a subaccount setup. Just create an internal extension on their server, and this will generate a SIP URI, e.g. 123456666@sip.us4.voip.ms where 12345 is your voip.ms account number and 6666 is the internal extension you created. This lets you connect directly with your server through the SIP URI from anywhere once you map this subaccount to an extension or IVR on your server. The charge for SIP URI calls is only $.001 per minute. The last step is to use this SIP URI in your remote SIP phone to connect back to your server. You can take advantage of the full range of Asterisk functions once these calls reach your server including IVRs and DISA. The approach is not only simple to implement, but it's also safe and economical.

There are some other alternatives as well. Use something like Google Voice or Ooma to redirect calls to your cellphone when you're traveling. Or buy an Ooma for Grandma or a MagicJack for Joe College. These options also are safe, secure, and quite inexpensive.

Just Released: Remote Phone Meets Travelin' Man

Activating Inbound SIP on Your Server. If you still are hell-bent on opening SIP access to your server, the Incredible PBX already is preconfigured to support it. Just map the SIP ports on your hardware- based firewall to your server (UDP 5000:5082 and UDP 10000:20000). Once activated, anyone can reach you through the following SIP URI using the actual public IP address of your server: mothership@12.34.56.78. You also can adjust the e164 trunk in FreePBX to route inbound calls to any destination desired. Then register your phone number on e164.org and others can call you at no cost using your traditional phone number. Enjoy!


The Incredible PBX: Basic Installation Guide

Adding Skype to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Multiple Google Voice Trunks to The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Basic Installation Guide, Part II.

Continue reading Basic Installation Guide, Part III.

Continue reading Basic Installation Guide, Part IV.

Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with the latest Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won't have to wait long for an answer to your questions.




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...

  1. We, of course, continue to recommend a dLink Router/Firewall. Low Cost: $35 WBR-2310  Better: DIR-825  Best: DGL-4500 []
  2. We recommend the free SipAgent client for Android devices and the commercial Acrobits Softphone for iPods and iPads. []
  3. Adding an entry like the following would dramatically reduce the likelihood of a SIP attack: -A INPUT -p udp -m udp -s 141.146.0.0/255.255.0.0 --dport 5000:5082 -j ACCEPT []

Orgasmatron 5.2: The Secure Swiss Army Knife for Asterisk

It’s been an exciting couple of weeks watching the overwhelmingly positive response to our release of Orgasmatron 5.1. With this version, we introduced a new Asterisk® security model that took into account the ever-increasing security risks posed by exposing web and telephony servers to direct Internet access. The bottom line is this. If your telecom requirements still can be accomplished by placing a server securely behind a $35 hardware-based Internet firewall with no Internet exposure, then it makes absolutely no sense to dangle such a tempting target in front of the world’s most nefarious creeps.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Our experience suggests that the only trade off with this new approach is the inability to receive anonymous SIP calls… a small price to pay considering the potential financial and computer risks involved. You still can place outbound VoIP calls as well as placing and receiving calls using any of the phone numbers registered on your new PBX in a Flash server. And, thanks to Google Voice, SIPgate, and IPkall, all inbound calls are free, and all outbound calls to numbers in the U.S. and Canada are free as well.

If a SIP URI and your own Freenum/ISN number are simply features you can’t live without, sign up for a voip.ms IAX account, and you’ll get a SIP URI for free. Inbound SIP URI and Freenum/ISN calls will set you back $1 for every 1,000 minutes billed in 6 second increments.

Or you can sign up for a free IP Freedom CallCentric account and configure a new SIP trunk in FreePBX by following these directions. Once configured, your new server SIP URI will be 1777xxxxxxx@in.callcentric.com where xxxxxxx is your assigned 7-digit CallCentric number.

Keep in mind that a new security vulnerability has been found with either Asterisk or FreePBX almost monthly. The chart below tells you why. With virtually limitless attack surfaces because of the number of interrelated components in CentOS, Asterisk, and FreePBX comes enormous and recurring potential for remote compromise of these systems. Rather than play this cat-and-mouse security game with the underworld, the Orgasmatron design changes the paradigm. It lets you use any (secure or insecure) version of Asterisk and FreePBX without worrying about any outside attacks. Do passwords on your new server matter? Not really… unless there is someone inside your firewall that you don’t trust. :roll: Are we going to secure them anyway? Absolutely. But instead of the constant worry over new security vulnerabilities, Orgasmatron 5.2 lets you enjoy exploring the world of Asterisk and VoIP telephony with an incredibly rich feature set that you won’t find anywhere else, period! We’ll resist making any other device analogies, but the idea here is to protect the good guy (you!) while keeping the bad guys out. No penetration. No worries. Simple as that.

In our former life working for a living, we actually procured and managed multimillion dollar PBXs as part of our “other duties as assigned.” Without qualification, we can tell you that the feature set that Orgasmatron 5.2 brings to the table for free runs circles around anything you could buy (then or now) in the commercial marketplace. And, at one time or another, we purchased every Nortel feature good money could buy. There’s one other difference. Orgasmatron 5.2 runs swimmingly on a $200 Atom-based PC that you can purchase at any Best Buy as well as hundreds of other stores including Amazon, NewEgg, and Buy.com. We paid more than $200 to provision an additional extension on our Nortel switch! You, of course, can add as many extensions as you like. De nada.

So, why a new version of Orgasmatron in only a few weeks? Well, it’s not security-related. In fact, there is nothing wrong with continuing on with Orgasmatron 5.1. Unfortunately, it relied exclusively upon SIPgate to make free Google Voice calls in the U.S. and Canada. And SIPgate required an invite using an SMS message from a U.S.-based cellphone. That pretty well knocked out all of our friends living outside the United States. Today’s version fixes that by letting anyone sign up for a free IPkall phone number in Washington state. All you need is a valid email address. The setup process is a bit more complex because IPkall doesn’t support registered connections to their servers. But we’ll walk you through the additional steps and, once completed, your server will be just as secure as the SIPgate approach we set up with Orgasmatron 5.1. And few, if any, Linux skills are required to set up or manage Orgasmatron 5.2. As we’ve noted previously, if you can handle slice and bake cookies, you’ve got the necessary skillset! Be aware this is about a one-hour project, and you need to track through the article carefully, or the entire house of cards comes down.

New Asterisk Security Model. Orgasmatron 5.2 maintains our design goal of running an absolutely secure Asterisk PBX from behind a hardware-based firewall with either NO INBOUND PORTS exposed to the Internet with SIPgate or an IP-address-restricted IAX port for IPkall. Don’t defeat this security mechanism by exposing additional ports on your PBX in a Flash server to Internet access. And choose your NAT-based firewall/router carefully. All of these devices are not created equally. Not only do some perform better than others, but certain models are notoriously bad at handling NAT-based routing tasks, a critical requirement in the Asterisk VoIP environment. In almost every case of problems with one-way audio, the real culprit can be traced back to a crappy router. For $35, you really can’t go wrong with the dLink WBR-2310. If you want traffic shaping functionality as well, take a look at dLink’s Gaming Router, our personal favorite.

As long as your router, Google Voice, SIPgate, and IPkall passwords are secure, you can sleep like a baby. We use an intermediate SIP provider for Google Voice to set up free outbound Google Voice calls in the U.S. and Canada because Google Voice actually places two calls to connect you to your destination. First, you get a call back. And then the party you’re calling is connected. The SIPgate or IPkall trunk is used by Google Voice to call you back so the inbound call is always free. We handle the interconnection magic with Asterisk transparently so your calls appear to be processed as if you were using a standard telephone to dial out. Just refrain from using extension 75 in Asterisk for personal conferencing!

The choice is yours. You can use SIPgate with no incoming ports exposed to your server from the Internet. Or you can use IPkall and map UDP port 4569 (IAX2) on your hardware-based firewall to the internal IP address of your new PBX in a Flash server. Even with the IPkall setup, we’ve locked down IPtables (our Linux firewall) to restrict IAX access to several specific IP addresses so your server remains absolutely secure. We’ve also included support for FonicaTec’s IAX offering for those that want a backup IAX provider. We’ll have much more to say about IPtables in coming weeks.

If you’ve already installed Orgasmatron 5.1 and it’s working for you, do you need to upgrade? NO. With the exception of the new IAX support for IPkall, the code in Orgasmatron 5.2 is identical.

We, of course, continue to recommend that you sign up with Vitelity so you have an alternate communications vehicle in the event of a problem with your free service. Vitelity also can provide 911 emergency service for your home or home office. You can save a little money while supporting the PBX in a Flash project by using the links at the end of this article.

Swiss Army Knife Inventory. There’s no need for a Swiss Army Knife if you don’t know what all the blades are for. So, for those that are wondering what’s included in the Orgasmatron 5.2 build, here’s a feature list of the components you get in addition to the base PBX in a Flash build with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using the scripts that are provided.

Prerequisites. Here’s what you’ll need to get started:

  • Broadband Internet connection
  • Rock-solid NAT router/firewall. Recommend: $35 dLink WBR-2310
  • $200 PC on which to run PBX in a Flash or a Proxmox Virtual Machine
  • Free Google Voice account (HINT: Under $2 on eBay)
  • Free SIPgateOne residential account (Use cell to get SMS invite) OR
  • Free IPkall IAX account

Learn First. Install Second. Even though the installation process is now a No-Brainer, you are well-advised to do some reading before you begin. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. Then read our PBX in a Flash and VPN in a Flash knols. If you’re still not asleep, there’s loads of additional documentation on the PBX in a Flash documentation web site.

Today’s Drill. The installation process is straight-forward, but a little different than the Orgasmo 5.1 scenario because of the need to accommodate IPkall. Just don’t skip any steps. In a nutshell, here are the 6 Steps to Free Calling and an incredibly versatile, preconfigured Asterisk PBX:

1. Install the latest version of PBX in a Flash
2. Run the Orgasmatron 5.2 Installer
3. Configure a softphone or SIP telephone
4. Configure Providers for Orgasmatron 5.2
5. Enter your Google Voice and SIPgate/IPkall credentials
6. Change existing passwords to secure your system

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash installed. We recommend you install the latest PIAF 1.6 beta on a new Atom-based PC. This beta is virtually identical to version 1.4 except it uses CentOS 5.4 instead of CentOS 5.2. This means it works better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use the identical payload for versions 1.3, 1.4, 1.5, and 1.6 of PBX in a Flash. The beta label simply means we haven’t had time to sufficiently test CentOS. But this is not a Microsoft-style beta so fear not!

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Burn the ISO to a CD. Then boot from the installation CD and type ksalt to begin.

WARNING: This install will completely erase, repartition, and reformat ALL disks on your system! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can’t find the kickstart file. Just tab to OK and press Enter. Don’t change the name or location of the kickstart file! This will get you going. Think of it as a CentOS ‘feature’. :-)

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS, it will reboot. Remove the CD promptly. After the reboot, choose A option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes

When prompted, change the ARI password to something really obscure. You’re never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source on the Orgasmatron builds because of parking lot issues in the latest releases of Asterisk.

Running the Orgasmatron 5.2 Installer. Log into your server as root and issue the following commands to run the Orgasmatron 5.2 installer:

cd /root
wget http://pbxinaflash.net/orgasmo52.x
chmod +x orgasmo52.x
./orgasmo52.x

Have another 15-minute cup of coffee. It’s a great time to consider a modest donation to the Nerd Vittles project. You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Now run passwd-master1. Set your FreePBX passwords to something very secure but different from your Linux root password.

Next, type status2 and press Enter. Write down the IP address of your new server.

If you’re using IPkall, now’s the time to log in to your hardware-based firewall/router and map UDP port 45693 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don’t worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you’re using SIPgate instead of IPkall.

For good measure, we recommend you reboot your server at this point. The command to type is simple: reboot4

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone, and you’ll find lots of recommendations on Nerd Vittles. For today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 82812661 as the password for extension 701 and the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Don’t Forget! After you change your extension passwords later in this tutorial, you will need to update the password entry in X-Lite, or you will no longer be able to place calls! In fact, you will get locked out of your server for 90 minutes after three failed password attempts. So put this on a sticky note so you don’t forget, or you’ll regret it in about 15 minutes.

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. And there’s really no reason you can’t use both if they’re available in your location. Do NOT use special characters in your provider passwords, or nothing will work! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we’d recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You’ll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don’t worry. You can erase your cellphone number from your account once it is set up. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn’t matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you’ll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You’ll need these in a few minutes to configure PBX in a Flash. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you’ve opted to use IPkall, here’s the drill. First, you’ll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we’ll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don’t have an available number, your request apparently goes in the bit bucket. You’ll know because IPkall typically turns these requests around in a few minutes. Don’t worry about the mothership entry. We’ll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it’s dynamic, then you’ll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you’ve got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you’ll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It’s worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

We’re going to be making some entries in FreePBX which is the web-GUI that manages PBX in a Flash. For now, we simply need to enter your new IPkall phone number so that incoming calls to your IPkall number will actually ring on your softphone. Later, we’ll make some further adjustments once we get Google Voice humming along.

Using a web browser from your desktop, log in to FreePBX 2.6 at the following link substituting your server’s private IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master).

When FreePBX loads, choose Setup, Trunks, ipkall (iax). In the USER Context field, enter your 10-digit IPkall phone number. Click Submit Changes, Apply Configuration Changes, Continue with Reload to save your settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it’s a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Now let’s test your new phone number. Call your IPkall number from a cellphone or some other phone. Your softphone should ring. Answer the call, and be sure you have voice in both directions! Do not proceed without success here, or the rest of the adventure is a waste of your time.

Configuring Google Voice. Google Voice still is by invitation only so the first thing you’ll need is an invite. If you’re in a hurry, then stroll over to eBay where you’ll find lots of them for under $2. Once you have your invite in hand, click on the email link to set up your account. After you’ve chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type. Trust us.

Google then will place a call to your number and ask you to enter a confirmation code that’s been provided. When your cellphone (SIPgate) or softphone (IPkall) rings, answer it and punch in the number. Wait for confirmation. Then hang up.

As we mentioned earlier, there’s no reason you can’t set up both SIPgate and IPkall forwarding numbers in Google Voice. Just repeat the drill with the other provider’s number if you wish to activate both numbers for use with Google Voice. They’re not both going to ring simultaneously as you will see in a minute.

While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Finally, place a test call to your new Google Voice number and be sure your cellphone or softphone rings. Don’t move forward until you’ve been able to successfully place a call to your phone by dialing your Google Voice number. Once this is working, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to PBX in a Flash. We’re ready to insert your Google Voice credentials and SIPgate/IPkall number into PBX in a Flash. You’ll need four pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 11-digit SIPgate or IPkall RingBack DID (beginning with a 1). Don’t get the 10-digit GV number mixed up with the 11-digit SIPgate/IPkall RingBack DID, or nothing will work. :-)

Log back into your server as root and issue the following command: ./configure-gv. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!!

Configuring FreePBX. Now shift back to your Desktop and, using a web browser, log in to FreePBX 2.6 at the following link substituting your actual IP address for ipaddress: http://ipaddress/admin. You’ll be prompted for a user name (maint) and password (the one you just created with passwd-master). Depending upon which intermediate provider you’re using, do the following:

SIPgate Setup. When FreePBX loads, choose Setup, Trunks, sipgate. In Peer Details, replace both instances of sipID with your actual SipGate SIP ID. In Peer Details, replace sipPassword with your actual SipGate SIP Password. In Register String, replace sipID with your SipGate SIP ID, replace sipPassword with your SipGate SIP Password, and replace 3333333333 with your 10-digit SipGate Phone Number. When finished, the Register String should look something like the following:

7004484f0:B8TTW3@sipgate.com/4155201234

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

SIPgate and IPkall Setup. While still in FreePBX with your browser, click Setup, Inbound Routes, gv-ringback. In DID Number, replace 3333333333 with your 10-digit SIPGate or IPkall Phone Number. In CallerID Number, replace 7777777777 with your 10-digit Google Voice Number.

Click Submit, Apply Configuration Changes, Continue with Reload to save your changes.

Securing FreePBX. You’re almost done. While still in FreePBX, choose each of the 16 preconfigured extensions on your new server and change the extension AND voicemail passwords. Here’s the drill: Setup, Extensions, 501, Submit. After changing secret and Voicemail Password, repeat with the next extension number instead of 501. Then Apply Config Changes, Continue when you’ve finished with all of them.

Now change the default DISA password: Setup, DISA, DISAmain, PIN, Submit Changes, Apply Config Changes, Continue.

Don’t forget to adjust your X-Lite password to match the password entry you made for extension 701!

Orgasmatron Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let’s place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. If everything is working, congratulations!

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. Save the file and restart Asterisk with the command: amportal restart.

Choosing a VoIP Provider. For this week, we’ll point you to some things to play with on your new server. Then, in the subsequent articles below, we’ll cover in detail how to customize every application that’s been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we’d recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask.

The VoIP world is new territory for some of you. Unlike the Ma Bell days, there’s really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O – Nerd Vittles Orgasmatron Demo (running on your PBX)
  • 1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 – Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 – Schedule a hotel-style wakeup call from any extension
  • 1061*1061 – PBX in a Flash Support Conference Bridge
  • 882*1061VoIP Users Conference every Friday at Noon (EST)


Click above. Enter your name and phone number. Press Connect to begin the call.


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and the Orgasmatron build adds a bunch of additional options. See if you can find all of them. For starters, you’ll want to activate CallerID Lookups in FreePBX. Choose Setup, CID Superfecta, Default and enter the maint password you created with passwd-master. Then choose Tools, Module Administration, CallerID Lookup, Enable, Process and Save the Settings. Then edit each of the Inbound Routes and choose CallerID Superfecta as the CID Lookup Source. Save your changes. Finally, choose Setup, CallerID Lookup Sources, CallerID Superfecta and be sure your maint password created with passwd-master is correct here, too. If not, update it. For additional tips, visit the forums.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You’ll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, if you’ve heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It’s also perfect for off-site backups!

Also check out Tweet2Dial which lets you use Twitter to make Google Voice calls, send free SMS messages, and manage your new Asterisk server. Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.

Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! We maintain a thread with the latest Patches for Orgasmatron 5.1 and 5.2. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. So you won’t have to wait long for an answer to your questions.

Coming Attractions. In our next episode, we’ll walk you through the process of adding a second, third, fourth, and fifth Google Voice line to your server so that you’ll never run out of free calling on your server. 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…

  1. passwd-master is the PIAF utility for setting a master password for FreePBX access with the maint user account. []
  2. status is the PIAF utility program that displays the current status of most major applications running on your server. []
  3. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device “inside” your firewall for further processing. []
  4. reboot is the Linux command for restarting your server. It’s functionally equivalent to shutdown -r now. []

Is It Time to Get Your Buzz On?

If you’ve been on vacation for the past week without your computer and cellphone, welcome back. Here’s what you missed: Everything! Yes, in seven little days, we’ve watched in amazement as Google the Giant Internet Gorilla with unlimited financial resources jumped off the 1,000 foot cliff without a parachute and then set about discovering how to land safely. Google jumped head first into social networking with a new product, Google Buzz, and decided to kick start the project using the age-old business trick of tying the new product to an old one, in this case their 175 million existing Gmail customers.

In the process of introducing Google Buzz, Google got almost everything wrong. Ironically, it’s the only new Google product that’s ever been introduced without the beta label. We’ll let you be the judge of whether Buzz is ready for prime time. Consider the following. There was the exposure of people’s most cherished Little Black Books to complete strangers. And cross-scripting security issues have been reported with the potential of exposing users’ Google credentials which in turn provides the key to the Google Checkout castle not to mention all of your most confidential emails. Then there were complaints that customer’s geolocation data was being compromised without user authorization. If that wasn’t enough on your plate for one week, Google now has to contend with a class action lawsuit and several government investigations into its Buzz business practices. Aside from that, did we mention Google Buzz is a huge hit!

If social networking is your thing, then you’re going to love Google Buzz. Think of it as FriendFeed on Steroids. Rich multimedia and location-based services on top of everything you always loved about IRC. And, if you have an Android 2.0+ phone, it gets even better with complete integration into Google Maps 2.0.

Unfortunately, everyone has been so busy with damage control and Google bashing that there’s been precious little time to actually explore the potential for Google Buzz in the social networking community. Our brief look at the product and its potential suggests that Google has another winner on its hands. It’s just too bad it wasn’t introduced in a manner similar to Google Wave so that users (and Google) could walk before attempting to fly. Here’s our first crack at how Google Buzz could actually be integrated into the blogging community, in our case making Google Buzz an integral part of a WordPress blog. And there’s also the widget approach from MoreTechTips.net which we’ve tweaked in the right margin below our Google maps. We’ll have more to say about these methodologies in coming weeks. In the meantime, come join the fun and Get Your Buzz On.

Nerd Uno’s Latest Buzz



Some Recent Nerd Vittles Articles of Interest…

Welcome to IP Country: A New Layer of Asterisk Security

image courtesy of fail2ban.org One of the problems with writing a blog like Nerd Vittles is it's more than double the work of your typical blog where a writer pontificates about something and then moves on. What makes Nerd Vittles a little different is that, with help from a number of very gifted developers, we actually create useful applications and then write about how to use them. So you get a bonus for the same low price: free! This obviously imposes some time constraints in order to get fresh material into your hot little hands every week.

This week we turn our attention to Asterisk® Security again and unfortunately the Whole Enchilada is not yet ready. So today you get Chapter I of this topic with a comment that we're still mulling over some enhancements. When those pieces are finished or at least properly evaluated, we'll produce a sequel. Software houses spend years developing applications. And sometimes it takes us more than a week. :-)

Let's start with a few observations which should be quite obvious to those who have wrestled with VoIP or Asterisk for a while. Internet security is a bitch. And Asterisk security is much, much worse. When a few disgruntled people can bring Twitter to its knees because they're mad about some particular tweet or Twitter user, it tells you what we're all up against. Hate to say it but we can all thank Microsoft for years of security neglect that rendered the Windows operating system less than optimum in preventing the spread and deployment of BOTs. And the tools have gotten more dangerous as well. Strangers (our euphemism for these folks) write new software, too.

If you're using PBX in a Flash (and you really should be!), you know that we've devoted enormous resources to Asterisk security. Two years ago when PBX in a Flash was introduced, the majority of people using Asterisk still were using 1234 as the extension password on all or most of their extensions. A couple $100,000 phone bills and lots of public education, and that situation hopefully is behind us. Two years ago, no Asterisk aggregation included a firewall... except PBX in a Flash. Believe it or not, there were individuals running Asterisk servers on the public Internet with a default root password of password. That added more than a few more BOTs to the Internet kettle of fish. Then there were the brute force password hacks that hit Asterisk servers thousands of times per minute guessing passwords. Nothing stood in the way of these attacks until PBX in a Flash introduced Fail2Ban which automatically blacklisted IP addresses after a certain number of failed login attempts. We followed Fail2Ban with our Atomic Flash product which provided a turnkey Hamachi VPN implementation for rock-solid safe remote computing. And, of course, there was a one-minute Hamachi VPN install script for standard PBX in a Flash systems. No other aggregation has it to this day.

The purpose of the history lesson isn't to crow about PBX in a Flash although we're mighty proud of it. Rather we wanted to make you aware that precious little development effort is actually going into security while enormous resources are devoted to things such as Internet faxing, Skype, and Google Voice integration. We'll be the first to admit that we love the latest gee whiz gizmos as much as anybody. But come on. A handful of us who do this purely for fun somehow manage to turn out loads of security enhancements while huge, for-profit companies are devoting virtually zero resources to making Asterisk, SIP, and the VoIP community safer. SIP is about as secure as whispering at a movie theater. Google releases Google Voice with SIP access protected by a 4-digit password. :roll: That approach to security needs to change, or we're all going to wake up sorry one day soon. If this is preaching to the choir, then feel free to pass this article on to one of your brethren who has not yet seen the light! Start by reading our Primer on Asterisk Security.

If you have extremely secure passwords on your Asterisk extensions and trunks, and you have deployed a properly configured firewall with Fail2Ban to protect against brute force attacks, then you're ahead of the curve insofar as Asterisk security is concerned. But what we think is still missing is access restrictions based upon what the military calls a "need to know." Simply stated, it means folks shouldn't get access of any kind to your Asterisk server unless they have a need to be there. And, if we find someone there that doesn't belong, they should be kicked off and banned from further access.

So today we have a new security tool for your Asterisk toolbox: IP Country, country-based network filtering by IP address. In a nutshell, it means configuring your Asterisk server to dramatically reduce the number of IP addresses which can reach your server at all. If you receive anonymous SIP connections from all around the globe that you actually need or if you're attacked from a BOT running on grandma's Windows machine down the block, this may not work for you, but it's another tool in your quiver of arrows. For most servers, it has the potential to reduce the vulnerability from random outside threats substantially. It's taken a lot of research to come up with much of what follows, and we want to express our special thanks to Sandro Gauci and Joe Roper for their assistance. Some of this technology has been around for many years, but unfortunately it was expensive. So we also want to express our special appreciation to MaxMind for releasing their open source GeoLite Country database which is now free for downloading. That is the critical ingredient in much of what follows. So here's a word from our sponsor:

This product includes GeoLite data created by MaxMind, available from http://www.maxmind.com/.

Scope of Protection. An obvious question is just exactly what are we trying to protect. In our view, it's several things. First, we don't want strangers logging in to extensions on our server and making free calls around the globe using pilfered or hacked passwords. We also don't want strangers using our extensions to masquerade as us for any other purpose. Second, we don't want strangers randomly calling our server using SIP URI's that they've dreamed up. And third, we don't want strangers accessing any other applications on our server including SSH and FTP as well as web and email services.

IP Country Design. As with other security features in Asterisk, FreePBX, and IPtables, our implementation of IP Country uses permit and deny access tables that consist of authorized and unauthorized ranges of IP addresses. There's also a table with the latest GeoLite Country information which is used as the data source for your permit table. When a connection to the server is made, the IP address is checked against the permit table of authorized addresses. If there's no match, we'll consider the connection a stranger. If there is a match, then we'll check the deny table to make certain this particular IP address hasn't been banned. Unless you alter all of our scripts, your system must be using the default MySQL account name of root with a password of passw0rd. As configured in PBX in a Flash, this is NOT a security risk since MySQL access is limited to your server, and your server requires root credentials to log in.

Today's Objective. To get everyone started, we're going to tackle the first two objectives today. The solutions offered should work fine on any FreePBX-based Asterisk system... even those that hide the existence of FreePBX.

For outgoing calls, we'll introduce a new script which runs periodically to examine the IP addresses attached to every SIP and IAX extension and trunk on your Asterisk server. If a stranger's IP address is identified (as explained above), we'll add an IPtables firewall rule to permanently block access to your server from this IP address. These rules are stored in /etc/sysconfig/iptables should you ever need to remove an IP address that has been blocked. You can adjust the script execution frequency based upon the thickness of your wallet. After all, it's your phone bill. This functionality is mutually independent from the incoming call protection outlined below so you can use either or both of the functions to meet your own requirements. For systems that use enormous numbers of SIP URI's for communications around the globe, you might choose to implement just this piece for extension and trunk IP Country protection without altering your incoming dialplan at all. Keep in mind that FreePBX now supports permit and deny IP address filters on extensions, something you really should be using even if you decide against implementing the IP Country security protection layer.

For incoming calls, we're going to modify FreePBX's existing Blacklist functionality to also look up the calling IP address in our IP Country permit and deny tables. If the IP address is authorized, the call will go through. Otherwise, the call will be treated just as if the caller's number were blacklisted. Be aware that incoming calls to one of your commercial DIDs may reflect the IP address of your provider since the caller may be calling from a Plain Old Telephone rather than an IP address. The existing Blacklist functionality can be used to block these unwanted callers. If you live in the United States, you'll probably also want to call 888-382-1222 and place your DIDs in the Do Not Call database. Just call from a phone using the CallerID of the number you wish to block.

Installing GeoLite Country. To get started, log into your server as root and issue the following commands:

cd /
wget http://bestof.nerdvittles.com/applications/ipcountry/ipcountry.tgz
tar zxvf ipcountry.tgz
rm ipcountry.tgz
cd /root/ipcountry
./nv-ipcountry

Once the nv-ipcountry script begins to run, it will download and install the GeoLite Country database into MySQL. You then will be asked whether to add countries to your permit table. Since your permit table is empty at this point, the answer should be yes. You'll then get a list of country codes. Choose the two-character country code desired and type it in UPPERCASE, e.g. US. If you want to add one or more additional countries, just rerun ./nv-ipcountry and do NOT initialize the permit table (which erases all of its contents).

New GeoLite Country databases are released every month or two so get used to the procedure. You'll be using it periodically to keep your list of IP addresses current. We'll cover the update procedure after we get you up and running.

Remember: If no IP addresses for any country are added to the permit table, you will not be able to make calls or register trunks with your providers! The only default entries added to the permit table are the non-routable, private IP address ranges, e.g. 192.168.0, etc. The geolite table is merely a data repository of the latest GeoLite Country database and has no effect on the daily operation of your system! You use it only as a data source for populating your permit table.

Testing IP Country. Before we actually turn anything on, we need to be sure we're not going to blow your Asterisk system out of the water! In short, we want to make sure that every extension that's supposed to be able to make a connection to your PBX still can. And we need to make sure all of your trunk registrations still are working. While you're still in the /root/ipcountry directory, issue the following command: ./test.sh. This script will display all of your SIP and IAX connections and then will tell you whether each connection will pass muster with IP Country security in place. Each IP address should display ok. If any of them show ko, you have a problem. This means that you have an extension or trunk with an IP address that is not included in your permit table. You can scan through the show peers listings in the display to figure out which providers or extensions are associated with any problem IP addresses. Be sure it's not a bad guy first. Then you have a couple of options. You can either manually add the IP address to the permit table as outlined below. Or you can add additional countries which include the missing IP address(es). To decipher the country of any problem IP address, go to this link and plug in the IP address. Once you've made entries in your permit table to cover all of your needed IP addresses, run the test script again just to be sure everything shows ok. Do NOT proceed until you get all ok's, and don't write us if you do.

Manually Adding IP Addresses to IP Country. We've provided a command-line utility which makes it easy to add IP addresses and address ranges to either the permit or deny tables of IP Country. Be very careful using this tool! There's limited error-checking which means it's easy to create a mess. You'll find iputility.php in the /root/ipcountry folder. Since all IP addresses are stored as integers, you can use it to merely discover the integer value of an IP address, or you can actually insert IP addresses into either the permit or deny tables. Here are a few examples to show how the utility works:

./iputility.php 156.130.20.10
Returns the integer value for this IP address; no database update
./iputility.php 156.130.20.10 156.130.20.255
Returns integer values for this IP address range; no database update
./iputility.php 156.130.20.10 deny
Adds this IP address to IP Country deny table
./iputility.php 156.130.20.10 156.130.20.255 permit
Adds this address range to IP Country permit table)

A couple of points worth noting. First, all custom entries in your permit and deny tables using iputility will show a country code of AA. This makes them easy to find using phpMyAdmin if you make a mistake. Second, if you attempt to enter the same IP address range more than once, you'll get a database error since all entries in the tables must be unique. Third, remember that entries in the deny table take precedence over entries in the permit table. So, if the same IP address or address range is in both tables, access will be denied. The reason for this is to make it easy to exclude a few bad apples from a country that you might otherwise find unobjectionable. Finally, keep in mind that manual entries added to the permit table will have to be added again each time you initialize the table and insert new country IP codes after a GeoLite Country refresh. The deny table is unaffected by database refreshes. So make yourself a list of entries you manually insert into the permit table and keep it in a safe place for future reference.

Activating the IP Address Checker. In the /root/ipcountry directory, you'll find the script that we'll use to check your system periodically to be sure all of the extensions and trunks are registered at permitted IP addresses. To run the script manually, log into your server as root and type: /root/ipcountry/ip-checker.sh. When you run it, you shouldn't see any modifications to IPtables, just a string of ok's. So now we want to added the script as a cron job that will be run periodically to watch your system. Edit /etc/crontab and insert the following line at the bottom of the file:

*/1 * * * * /root/ipcountry/ip-checker.sh > /dev/null

*/1 means run the script once a minute, all day and night, every day. */5 means every 5 minutes. You make the call on how safe you'd like your system to be. If you'd like to receive an email or text message every time an IP address is blocked by ip-checker.sh, just edit the filecheck.php script, uncomment the two lines that begin with // and replace yourname@gmail.com with your email or text message address.

WARNING: For ip-checker.sh to work properly with IPtables, there are a couple of prerequisites. First, IPtables must be running on your system with the iptables file located in /etc/sysconfig. Second, your IPtables setup must include an SSH permit rule that looks like this:

-A INPUT -p tcp -m tcp --dport ssh -j ACCEPT

We use this rule as a place finder to determine where to insert new rules to block stranger's IP addresses. If you don't have the above rule, filecheck.php (used by ip-checker.sh) won't be able to insert new rules. So you'll need to manually edit filecheck.php to provide a "hook" that can be used to insert rules into your iptables file. PBX in a Flash systems come preconfigured to support this. With other aggregations, YMMV!

Activating the Incoming Call Checker. To screen incoming calls using your IP Country permit and deny tables, the setup is straight-forward assuming you are running the latest version of FreePBX 2.5. We're going to adjust the Blacklist context to also perform IP address lookups from IP Country when new calls arrive on your PBX. Just log into your server as root and add the following lines to the bottom of the extensions_override_freepbx.conf file in /etc/asterisk:

[app-blacklist-check]
include => app-blacklist-check-custom
exten => s,1,LookupBlacklist()
exten => s,n,GotoIf($["${LOOKUPBLSTATUS}"="FOUND"]?blacklisted)
exten => s,n,Set(TESTAT=${CUT(SIP_HEADER(From),@,2)})
exten => s,n,GotoIf($["${TESTAT}" != ""]?hasat)
exten => s,n,Set(FROM_IP=${CUT(CUT(SIP_HEADER(From),>,1),:,2)})
exten => s,n,Goto(gotip)
exten => s,n(hasat),Set(FROM_IP=${CUT(CUT(CUT(SIP_HEADER(From),@,2),>,1),:,1)})
exten => s,n(gotip),NoOp(Gateway IP is ${FROM_IP})
exten => s,n,NoOp(IP Country Lookup in Progress...)
; put authorized special calls like sipgate's Google Voice ringbacks below
exten => s,n,GotoIf($["${FROM_IP}"="sipgate.com"]?keepon)
exten => s,n,AGI(nv-ipcountry.php|${FROM_IP})
exten => s,n,GotoIf($["${STRANGER}"="true"]?blacklisted)
exten => s,n(keepon),NoOp(** AUTHORIZED CALLER **)
exten => s,n,Return()
exten => s,n(blacklisted),Answer
exten => s,n,Wait(1)
exten => s,n,Zapateller()
exten => s,n,Playback(ss-noservice)
exten => s,n,Hangup

Make sure you remove the line-wrap in the s,n(hasat) line and any others that may have wrapped in the display above! Then save the file and reload your Asterisk dialplan: asterisk -rx "dialplan reload". You're all set! If you'd like email notices when a stranger calls and is blacklisted, edit nv-ipcountry.php in /var/lib/asterisk/agi-bin. Plug in your actual email address in the $email variable and set $emailalerts = 1.

Housekeeping 101. As we mentioned above, the pool and location of IP addresses continues to change so periodic updates are necessary, or you'll end up blocking calls that otherwise should be permitted. MaxMind updates GeoLite Country on the first day of every month so add it to your TO-DO list. We strongly recommend that you perform these steps through an SSH connection from a remote PC. Why? Because, if you forget step 1 while logged directly into your server, you could inadvertently lock yourself out of your own system if the ip-checker script happens to run while your permit table is empty. If you do it from a remote machine, you can simply move to another machine and follow these instructions properly. Otherwise, you've got a serious problem on your main server. If this server provides phones to your business, do the update when the server is idle. So here's the drill:

  1. Comment out the ip-checker.sh /etc/crontab entry
  2. Download new GeoLite Country database from MaxMind
  3. Initialize the ipcountry.permit table
  4. Add authorized countries back into ipcountry.permit table
  5. Add back any custom entries to permit table
  6. Test your IP Country system to make sure you get all ok's
  7. Reactivate ip-checker.sh in /etc/crontab

1. Log into your server as root. To comment out the ip-checker.sh line in /etc/crontab, just add # as the first character on the line and save the file.

2. Change to the /root/ipcountry directory and run ./nv-GeoIPrefresh.

3. While still in the /root/ipcountry directory, run ./nv-ipcountry and choose 1-Yes to initialize your ipcountry.permit table.

4. Continue running or rerun ./nv-ipcountry to add each desired country to your ipcountry.permit table.

5. Run ./iputility.php to add custom IP address entries to your ipcountry.permit table. You do NOT need to reenter addresses in the deny table. It is unaffected by this update procedure.

6. Test your system again to make sure all extensions and trunks get an ok by running ./test.sh.

7. Edit /etc/crontab and remove the # at the beginning of the ip-checker.sh line and save the file.

What's Next. We're still exploring another possibility with IP Country, and that is integrating GeoLite Country directly into IPtables. This would validate every packet coming into your firewall using IP Country-like rules in IPtables. If you want to look at how it could be done, see this excellent writeup. Well, not so fast. Unfortunately, it won't compile under CentOS 5.2. Here's a link to the problem code if there are any Linux gurus in the house. Our reluctance in doing this has to do with performance. Keep in mind that, without stateful packet inspection, every single packet coming into your server would presumably trigger a database lookup. On a busy telephony system generating hundreds of thousands of packets per second, it would take a beast of a server with sufficient memory to cache the entire IP Country database in order to handle the processing load. So now we've got to either learn about or find an expert on the IPtables State Machine. If anyone wants to experiment, please share your expertise with the rest of us. There's a Google Voice invite in it for you, too.


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