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The Poor Wise Man’s Burglar Alarm System with Asterisk: Under $10/month

Monday, July 14, 2014 / 7 Comments on The ~~Poor~~ Wise Man’s Burglar Alarm System with Asterisk: Under $10/month

If you’re like us, spending $50 a month or more on a home security system is a bit like pouring money down the toilet. Add to that the complications of getting one to work reliably with VoIP without spending another $50 a month on a Ma Bell vintage telephone line just adds insult to injury.

So perhaps you can share our elation when an email arrived last week announcing Straight Talk’s new Remote Alert System, a $10/month cellular-based system that uses Verizon Wireless to provide SMS and phone call alerts for up to eight numbers. And actually it’s cheaper than that. $100 buys you a year of service. That’s less than $8.50 a month. Today we’ll show you how to transform your Prius-like Remote Alert System into a Tesla that will rival virtually any intrusion detection system on the market… at any price! The extra hardware required: any Asterisk-based server including the Raspberry Pi and BeagleBone Black.

Read and weep, ADT!

If we didn’t already have three Straight Talk lines of service, we would have filed this in the Too Good To Be True pile and moved on. But we’ve had terrific Almost-Unlimited™ AT&T Wireless service with Straight Talk for less than $500 a year. It’s not only indistinguishable from AT&T’s own offerings costing at least 50% more, but it’s also contract-free so we can bring any AT&T smartphone including iPhones to the party and never miss a beat.

We decided to take the bait and ordered the home security bundle. This gets you the Remote Alert wireless controller plus a wireless motion sensor plus a year of service for $229.99. If you prefer a one-month gamble, the bundle is only $139.99. Down the road, you can add additional motion sensors and window/door sensors for about $30 each. The add-ons now are available at Wal-Mart.

Shameless Plug. We obviously don’t charge for access to our articles. But you can assist the Nerd Vittles project financially by using our referral link with eBates® to make your purchase if you decide to try this. It doesn’t cost you a dime but returns 13.5% of your purchase price to the Nerd Vittles project. It’s just a couple of clicks. Start here to access eBates. Then Search for Straight Talk and click on the link. After the Straight Talk web site displays, click on the following link to access the Straight Talk Security Bundle. And, THANK YOU!

So… back to our story. The controller supports four zones for monitoring. Zone 4 is reserved for sensors you want to monitor while someone may still be moving around in the house, for example while only some of your family may be sleeping or if pets are roaming. The other three zones typically would be used for motion sensors that trigger alerts when anything moves… after giving you 30 seconds to leave and return, of course. You can activate Home or Away monitoring using either the controller, an optional $25 key fob, or a free app for your iPhone or Android smartphone.

You get to decide what happens when the system is armed and an alert is triggered either by motion or a monitored door or window being opened. For us, silence was the name of the game. Using the Android Remote Alert System, click the Silent ARM icon once you leave the house, and you’re done. When you return, click the Disarm icon within 30 seconds of opening the door, and monitoring is disabled. You can also enter your 4-digit alarm code on the controller to disable monitoring.

Remote Alert System Setup. Once you get the equipment, it’s a 5-minute phone call to get set up. Install the backup batteries in the controller and motion detector, and plug the controller into an A/C power source. Press the required sequence on the controller to activate it, and you’re in business. The motion detector is already paired with the controller when it arrives, but adding new sensors is a 15-second task. All of the commands are documented in the manual which accompanies the system. But the tutorials also are available on line if you want to have a look.

Step #1 is changing your security alarm password. The next step is entering your phone numbers. Straight Talk goes to great lengths warning you that this is not a home security system because it has no external siren and can’t make 911 calls. They obviously haven’t heard of Asterisk®. 🙂 But let’s get through the standard setup before we talk about Asterisk integration. You get to set up three numbers to receive SMS text messages when an alarm is triggered. And you get to set up five phone numbers to receive calls when an alarm is triggered. What the called party will actually hear is an obnoxious alarm tone which continues to play for 15 seconds. If you had multiple properties with alarm systems and no Caller ID, you’d never know the source of the alarm! But people with multiple properties probably aren’t smart enough to use this system to begin with so let’s move on. You configure the SMS and phone numbers by entering a special code on the controller to program each of the eight destinations. Then you enter the 10-digit number twice, and you’re done. Easy Peasy!

If you’re new to home security systems, the key to motion sensors is placement. Straight Talk recommends placement about seven to ten feet off the floor with a wide field of view. The range of the motion sensor is about 26 feet. It obviously depends upon the layout of your house or apartment, but we had much better success placing the motion sensor on a window sill at about 5 feet high and aiming it at the center hall of our home. It improved the motion detection dramatically. Trial and error is your friend!

The next step is positioning your controller. A mounting bracket is included so that you can place it almost anywhere you like. Our preference is to hide it so long as it still has Verizon cellular coverage and a source of electricity. You can test it by arming the controller with your smartphone and then triggering the motion sensor. If you get an SMS message or a call, it’s working. We also prefer silent mode. An intruder is obviously going to attempt to destroy your controller if they hear it. Yes, the intruder may leave, but they’ll probably carry some of the family jewels with them. With an Asterisk server in place, we’d prefer to send the police without alerting the intruder that something has gone wrong.

Asterisk Integration. Speaking of Asterisk, here’s what we’ve developed to add 911 alerts and telephone alarms to this system. It’s a 5-10 minute project! The way this works is to first add a phone number to your controller that calls a dedicated DID on your Asterisk server. Calls to that DID trigger the special context [st-remote-alert] which verifies the CallerID number of your alarm system. As configured, if the CallerID doesn’t match, the call is immediately disconnected although you could easily modify our code to use an existing (non-dedicated) DID if you prefer. Just route the non-matching CallerIDs to whatever context you traditionally use to process inbound calls. If the CallerID of the alarm system is matched, then the call is disconnected AND an outbound call is placed to 911. When the 911 operator answers, a prerecorded message is played at least twice that says something like this using REAL information:

This is an automated security request for assistance from the residence at 36 Elm Street in Podunck, Arkansas. The owner of this residence is Joe Schmo at phone number: 678-123-8888. An intruder has been detected inside the home. A suspected burglary is in progress. All of the residents of the home are unavailable to place this call. Please send the police.

The phone number from which this automated call is being placed is 678-123-4567. If the owners have a working cell phone, you can reach them at the following number: 678-123-9999. Please dispatch the police to 36 Elm Street immediately, whether you can reach the owners or not.
A suspected burglary is in progress. Thank you for your assistance. This message will repeat until you hang up…

You can either use Flite and Igor to play the message, or you can record your own message to be played to 911. Use the FreePBX® Admin -> System Recordings option. We recommend the latter especially since you’ll be sending these emergency calls to 911. You obviously want the 911 operator to be able to quickly decipher what’s being said.

Legal Disclaimer. We cannot stress strongly enough that you need to test this carefully on your own server by placing test calls to some number other than 911 until you are positive that it is working reliably as determined solely by you. Be advised that this system will not work at all in the event of an electrical, Internet, or server outage. As delivered, this code will NOT place calls to 911. The choice of whether to modify the code to place 911 emergency calls is solely yours to make. Be advised that false and inadvertent calls to 911 may result in civil and criminal penalties. DON’T BLAME US!

NO WARRANTIES, EXPRESS OR IMPLIED, INCLUDING THE IMPLIED WARRANTY OF FITNESS
FOR A PARTICULAR PURPOSE AND MERCHANTABILITY, ARE BEING PROVIDED.

BY PROCEEDING WITH IMPLEMENTATION AND INSTALLATION OF THIS SOFTWARE, YOU AGREE
TO ASSUME ALL RISK AND COMPLETE RESPONSIBILITY FOR ANY AND ALL CONSEQUENCES
OF IMPLEMENTATION WHETHER INTENDED OR NOT AND WHETHER IMPLEMENTED CORRECTLY
OR NOT. YOU ALSO AGREE TO HOLD WARD MUNDY, WARD MUNDY & ASSOCIATES LLC, AND
NERD VITTLES HARMLESS FROM ALL CLAIMS FOR ACTUAL OR CONSEQUENTIAL DAMAGES.
BEFORE IMPLEMENTING AUTOMATED 911 CALLS, CHECK WITH A LOCAL ATTORNEY TO MAKE
CERTAIN THAT SUCH CALLS ARE LEGAL IN YOUR JURISDICTION.

IN THE EVENT THAT ANY OF THESE TERMS AND CONDITIONS ARE RULED UNENFORCEABLE,
YOU AGREE TO ACCEPT $1.00 IN COMPENSATION FOR ANY AND ALL CLAIMS YOU MAY HAVE.

THIS SOFTWARE IS FREE AND YOU AGREE TO ASSUME ALL RISKS WHETHER INTENDED OR NOT.
YOU ALSO ACKNOWLEDGE AND UNDERSTAND THAT THINGS CAN GO WRONG IN TECHNOLOGY.

WE CANNOT AND DO NOT WARRANT THAT THIS CODE IS ERROR-FREE OR THAT IT WILL
PROTECT YOUR PROPERTY, YOUR LOVED ONES, OR ANYONE, OR ANY THING IN ANY WAY.

IF YOU DO NOT AGREE WITH THESE TERMS AND CONDITIONS OF USE, DO NOT PROCEED!

Asterisk Implementation. First, you’ll need a dedicated DID that can be used to receive incoming calls from your Remote Alert System. Hopefully, you won’t be receiving many calls on this number so any of the inexpensive pay-by-the-minute DIDs will suffice. Or you can use a free DID from ipkall.com. The only gotcha with ipkall.com is having to make a call to keep the number active at least once every 30 days. But this could be accomplished with a weekly telephone reminder that only connected for a few seconds. Just don’t make the weekly call using the CallerID of your alarm system. You obviously do not want to trigger a 911 emergency call.

Next, you’ll need an outbound trunk on your Asterisk server that’s previously been registered with E911 support and that already is configured to place outbound 911 calls from your server. Google Voice trunks will not work! Your name, address, and phone number as they were registered with E911 will be important pieces of information to relay in your automated emergency call to 911. You’ll also need a cellphone number that can be provided with your 911 calls so that emergency responders have a way to contact you to follow up on automated emergency calls from your server.

Temporarily, you’ll also need a 10-digit number to which to deliver the automated emergency calls for testing. Your cellphone number would suffice. Once you’re sure everything is working, we’ll show you how to modify the dial plan code to replace this number with 911 when your system goes "live."

Installation. Once you have all of the required pieces in place, you’re ready to begin the installation. Log into your server as root and issue the following commands to begin:

cd /root
wget http://nerdvittles.com/wp-content/st-remote-alert.tar.gz
tar zxvf st-remote-alert.tar.gz
rm -f st-remote-alert.tar.gz
./st-remote-alert.sh

Once the install is finished, use FreePBX to modify the DID Trunk that will receive the incoming alerts from your Remote Alert System. Change the context entry to: context=st-remote-alert

Test. Test. Test. Testing is critically important before you actually turn on automated calls to 911. Once you’ve installed the software, activate your Remote Alarm System and then trip the motion detector to trigger a call to the dedicated DID on your Asterisk server. There’s typically a 30-second delay between tripping a motion detector and the commencement of the alert calls. Within a minute, you should receive a call on the emergency number you set up for testing. You can follow the progress of the procedure using the Asterisk CLI: asterisk -rvvvvvvvvvv. We recommend testing this repeatedly for at least a month before even considering 911 deployment. Make certain that everyone in your household knows how to disable the alarm system when they return home after arming it. Make certain that everyone in your household knows to never arm the system with motion detectors activated when anyone or any animal inside the house could potentially trip the alarm. At least until everyone is accustomed to these new security procedures and has a proven (successful) track record, NEVER DEPLOY SILENT ARMING OF YOUR REMOTE ALERT SYSTEM! If you change to silent arming of the Remote Alert System, test for at least another full month with no inadvertent failures before considering 911 deployment.

Making Changes. The st-remote-alert.sh installer has been designed to let you run it over and over again to replace or update your settings. So don’t be shy about making changes.

Substituting a Personally Recorded Message. If you’d prefer to record your own message to be delivered to 911, then review the script above and make yourself a cheat sheet before you begin. Then use a browser to open FreePBX. Choose Admin -> System Recordings and enter an extension number on your system to use for recording. Click the Go button to begin. Then dial *77 from that extension and record your message. Press # when you’re finished. Be sure to listen to the recording to make sure it’s what you intended. If not, rerecord the message until you get it right. You can dial *99 to listen to your recording a final time. When you’re sure it’s correct, name the recording nv-alert. Click Save.

Now you need to tell the automated alert dialer to use your recorded message instead of Flite and Igor.
Edit /etc/asterisk/extensions_custom.conf. Search for the line containing "pickrecording". Change Extension: 4 to Extension: 5. Save the file and reload your dial plan: asterisk -rx "dialplan reload"

Do some additional testing if you have substituted your own recording!

Adding Audible Alarms During Emergencies. If you prefer a little noise sprinkled around your home during burglaries, then we’ve put in place the necessary components to sound alarms on SIP phones that support AutoAnswer after feeding an extension to the speakerphone. For example, assuming you have deployed a Yealink T46G with an IP address of 192.168.0.10 and default admin credentials, you could add this additional line just before the final s,n,Hangup line in the [st-remote-alert] context of /etc/asterisk/extensions_custom.conf:

exten => s,n,System(curl -s -S --user-agent "Alert" http://admin:admin@192.168.0.10/servlet?number=25276)

To add additional Yealink phones, just add additional lines to the dialplan with the IP address of each phone. For other phone models, you’ll need to do a little research. 😉

Going Live with Automated Emergency Calls to 911. When you and everyone in your household are absolutely comfortable with the arming, disarming, and motion detection procedures, then you can decide whether to reroute the automated notifications to 911. Be advised that, in some states or municipalities, it may be illegal to auto-dial 911 from a non-human caller/system. Before doing this, check with an attorney or local authorities in your jurisdiction to make sure you are in compliance with federal/state/local laws.¹ If you elect to proceed, edit extensions_custom.conf in /etc/asterisk. Search for the line containing "SEND-HELP-REQUEST-TO". Replace the temporary number that you set up with the number: 911. Save the file and reload your dial plan: asterisk -rx "dialplan reload". Sleep well!

Originally published: Monday, July 14, 2014

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! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

Autodialers that make emergency calls to E911 as part of a burglar alarm system are specifically exempted in some states such as Illinois. This comports with federal law under The Telephone Consumer Protection Act (47 U.S.C. § 227). Emergency robocalls are specifically exempted from the new PSAP Do-Not-Call Registry rules. See also this article about E911 laws in the Northeast. In most cases, but not all, these laws target abuse of the E911 system. Surprisingly, one town that reportedly prohibits ALL autodialing to 911 is Palo Alto, CA. And Paris, Tennessee also has joined the illegal club. Special thanks to @TheMole on the PIAF Forum for his excellent research. [↩]

Knock Three Times: Pain-Free Remote Access to Your Asterisk or Linux Server

Monday, July 7, 2014

No. We’re not going to make you relive the 1970’s with us today although now you can listen to this Number 1 Hit and a million others for free with Amazon’s new Prime Music. No, we don’t get a commission if you sign up for Amazon Prime. Yes, we make millions when you buy something from Amazon using our links. Thank you! What we have for you today is a Number 1 Utility, and it works on virtually any Linux platform. If your fraternity or sorority had a secret knock to gain access, then you already know the basic concept. Port Knocker (aka knockd) from Judd Vinet is a terrific utility that runs as a daemon on your server and does just what you’d expect. It listens for knocks. When it detects three knocks on the correct three ports in the proper sequence and from the same IP address, it opens the IPtables Linux Firewall for remote access from that IP address to your server for a predefined period of time. This would allow you to log into your server with SSH or make SIP phone calls using a softphone registered to your remote Asterisk® server. What makes Port Knocker especially useful is the existence of knocking clients for virtually any smartphone, tablet, or desktop computer. For the Travelin’ Man, it’s another must have utility.

We introduced a turnkey implementation of Port Knocker in Incredible PBX for Ubuntu 14 late last week. If you were a pioneer earlier in the week, go back and install it again to take advantage of Port Knocker. Or better yet, follow along and we’ll show you how to install it on your own RedHat/CentOS or Ubuntu/Debian server in just a couple of minutes.

Prerequisites. We’ve built open source installation scripts for both the RedHat/CentOS platform as well as the Ubuntu/Debian operating systems. These knockd installers assume that you have a fully functional and locked down IPtables firewall with an existing WhiteList of authorized users. We’d recommend Travelin’ Man 3 if you need to deploy this technology and haven’t done so already. Last week’s Incredible PBX for Ubuntu 14 already includes Travelin’ Man 3 whitelisting technology. Read the article for full details.

Today’s knockd installers are fairly generic but, if you’re running a version of CentOS earlier than 6.x or Ubuntu earlier than 14 or Debian.anything, be advised that we haven’t tested these installers on those platforms so you’re on your own. Finally, if your server is sitting behind a hardware-based firewall (as we ALWAYS recommend), then you’ll also need to map the service you wish to access (e.g. UDP 5060 for SIP or TCP 22 for SSH) plus the three TCP ports from your hardware-based firewall to your server so that legitimate "knocks" can find their way to your server. The "knock" ports themselves do not need to be opened in your IPtables firewall configuration! We’re just knocking, not entering. 🙂

Overview. As configured, today’s installation scripts will install and preconfigure knockd to load automatically when you boot up your server. Three random TCP ports will be assigned for your server, and this port sequence is what remote users will need to have in order to gain access. Yes, you can change almost everything. How secure is it? Well, we’re randomizing the 3-port knock sequence using over 3,900 ports so you can do the math to figure out the odds of a bad guy guessing the correct sequence. HINT: 3900 x 3900 x 3900. Keep in mind that these "knocks" must all be received from the same IP address within a 15-second window. So sleep well but treat the port sequence just as if it were a password. It is! Once a successful knock sequence has been received, the default Port Knocker configuration will open all ports on your server for remote access from the knocking IP address for a period of one hour. During this time, "The Knocker" can log in using SSH or make SIP calls using trunks or extensions on the server. Port Knocker does not alleviate the need to have legitimate credentials to log into your server. It merely opens the door so that you can use them. At the bewitching (end of the) hour, all ports will be closed for this IP address unless "The Knocker" adds a whitelist entry for the IP address to IPtables during the open period. Yes, all of this can be modified to meet your individual requirements. For example, the setup could limit the range of ports available to "The Knocker." Or the setup could leave the ports open indefinitely until another series of knocks were received telling knockd to close the IPtables connection. Or perhaps you would want to leave the ports open for a full day or a week instead of an hour. We’ll show you how to modify all of the settings.

Server Installation. To get started, log into your server as root and download and run the appropriate installer for your operating system platform.

For RedHat/Fedora/CentOS/ScientificLinux servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-R.tar.gz
tar zxvf knock*
rm knock-R.tar.gz
./knock*

For Ubuntu/Debian servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-U.tar.gz
tar zxvf knock*
rm knock-U.tar.gz
./knock*

For ARM-based servers, issue the following commands:

cd /root
wget http://nerdvittles.com/wp-content/knock-ARM.tar.gz
tar zxvf knock*
rm knock-ARM.tar.gz
./knock*

Server Navigation Guide. On both the RedHat/CentOS/Fedora and Ubuntu/Debian platforms, the knockd configuration is managed in /etc/knockd.conf. Before making changes, always shutdown knockd. Then make your changes. Then restart knockd. On RedHat systems, use service knockd stop and start. On Ubuntu, use /etc/init.d/knockd stop and start. By default, knockd monitors activity on eth0. If your setup is different, on Ubuntu, you’ll need to change the port in /etc/default/knockd: KNOCKD_OPTS="-i wlan0". On RedHat, the config file to modify is /etc/sysconfig/knockd and the syntax: OPTIONS="-i venet0:0".

In /etc/knockd.conf, create an additional context to either start or stop an activity. It can also be used do both as shown in the example code above. More examples here. There’s no reason these activities have to be limited to opening and closing the IPtables firewall ports. You could also use a knock sequence to turn on home lighting or a sprinkler system with the proper software on your server.

To change the knock ports, edit sequence. Both tcp and udp ports are supported. seq_timeout is the number of seconds knockd waits for the complete knock sequence before discarding what it’s already received. We’ve had better luck on more servers setting tcpflags=syn. start_command is the command to be executed when the sequence matches. cmd_timeout and stop_command tell knockd what to do after a certain number of seconds have elapsed since the start_command was initiated. If you’re only starting or stopping some activity (rather than both), use command instead of start_command and stop_command to specify the activity.

IPtables 101. The default setup gives complete server access to anyone that gets the knock right. That doesn’t mean they get in. In the PIAF World, it means they get rights equivalent to what someone else on your LAN would have, i.e. they can attempt to log in or they can use a browser to access FreePBX® provided they know the server’s root or FreePBX credentials.

If you would prefer to limit access to a single port or just a few ports, you can modify command or start_command and stop_command. Here are a few examples to get you started.

To open SSH access (TCP port 22):

/sbin/iptables -A INPUT -s %IP% -p tcp --dport 22 -j ACCEPT

To close SSH access (TCP port 22):

/sbin/iptables -D INPUT -s %IP% -p tcp --dport 22 -j ACCEPT

To open a range of SIP ports (UDP 5060 to 5069):

/sbin/iptables -A INPUT -s %IP% -p udp --dport 5060:5069 -j ACCEPT

To close a range of SIP ports (UDP 5060 to 5069):

/sbin/iptables -D INPUT -s %IP% -p udp --dport 5060:5069 -j ACCEPT

Here’s a gotcha to be aware of. If you’re using the Travelin’ Man 3 WhiteList setup on your server, be especially careful in crafting your IPtables rules so that you don’t accidentally remove an existing Travelin’ Man 3 rule in closing some port with knockd. You will note that the syntax of the knockd commands is intentionally a bit different than what you will find in your Travelin’ Man 3 setup. This avoids clobbering something accidentally.

Monitoring Activity. Here are the two best tools to monitor knockd activity to make certain your setup is performing as expected. The knockd log (/var/log/knockd.log) will tell you when a knocking attempt has occurred and whether it was successful:[2014-07-06 14:44] starting up, listening on eth0 [2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 1 [2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 2 [2014-07-06 15:29] 79.299.148.11: opencloseSSH: Stage 3 [2014-07-06 15:29] 79.299.148.11: opencloseSSH: OPEN SESAME [2014-07-06 15:29] opencloseSSH: running command: /sbin/iptables -A INPUT -s 79.299.148.11 -p tcp --dport 22 -j ACCEPT

Next, verify that the IPtables command did what it was supposed to do. iptables -nL will tell you whether port 22 access was, in fact, enabled for 79.299.148.11. The entry will appear just above the closing Chain entries in the listing:

ACCEPT     tcp  --  79.299.148.11         0.0.0.0/0           tcp dpt:22

Two things typically can go wrong. Either the knock from a client computer or cellphone wasn’t successful (knockd.log will tell you that) or IPtables didn’t open the port(s) requested in your knockd command (the iptables -nL query will show you that). In the latter case, it’s usually a syntax error in your knockd command. Or it could be the timing of the knocks. See /var/log/knockd.log.

Port Knocker Clients. The idea behind Port Knocker is to make remote access easy both for system administrators and end-users. From the end-user perspective, the simplest way to do that is to load an app on the end-user’s smartphone so that even a monkey could push a button to gain remote access to a server. If the end-user’s cellphone has WiFi connectivity sitting behind a firewall in a hotel somewhere, then executing a port knock from the smartphone should open up connectivity for any other devices in the hotel room including any notebook computers and tablets. All the devices typically will have the same public IP address, and this is the IP address that will be enabled with a successful knock from the smartphone.

Gotta love Apple’s search engine. Google, they’re not…

There actually are numerous port knocking clients for both Android and iOS devices. Here are two that we’ve tested that work: PortKnock for the iPhone and iPad is 99¢ and PortKnocker for Android is free. Some clients work better than others, and some don’t work at all or work only once. DroidKnocker always worked great the first time. Then it wouldn’t work again until the smartphone was restarted. KnockOnD for the iPhone, which is free, worked fine with our office-based server but wouldn’t work at all with a cloud-based server at RentPBX. With all the clients, we had better results particularly with cloud-based servers by changing the timing between knocks to 200 or 500 milliseconds. How and when the three knocks are sent seems to matter! Of all the clients on all the platforms, PortKnocker was the least temperamental and offered the most consistent results. And you can’t beat the price. A typical setup is to specify the address of the server and the 3 ports to be knocked. Make sure you have set the correct UDP/TCP option for each of the three knocks (the default setup uses 3 TCP ports), and make sure the IP address or FQDN for your server is correct.

Another alternative is to use nmap to send the knocks from a remote computer. The knock.FAQ file in your server’s /root directory will tell you the proper commands to send to successfully execute a connection with your server’s default Port Knocker setup. Enjoy!

Originally published: Monday, July 7, 2014

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! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

The Definitive VoIP Quickstart Guide: Incredible PBX for the Raspberry Pi

Monday, June 23, 2014 / 5 Comments on The Definitive VoIP Quickstart Guide: Incredible PBX for the Raspberry Pi

It’s been a wild ride with the $35 Raspberry Pi®. Last month, sales of the Raspberry Pi topped three million. And, if you didn’t already know, the Raspberry Pi makes a near perfect platform for your very own VoIP PBX. It’s less than a ONE HOUR project!
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If you’re new to the party, imagine squeezing a 700 mHz ARM processor with 512MB of RAM, 2 USB ports, a 10/100 Ethernet port, an HDMI port, composite video, a separate audio jack, an SDHC card slot, and a micro USB port onto a motherboard the size of a credit card weighing 1.6 ounces. Adding WiFi is as simple as plugging in a USB adapter. If your WiFi connection is password-protected, just plug your credentials into the top of /etc/wpa.conf and /root/restart-wlan0.

07/01/2019 NEWS FLASH: Just released Incredible PBX LITE for the Raspberry Pi 2, 3, and 4 featuring Raspbian 10 Buster. Tutorial here.

Absolute perfection. Other than the slow write speeds to the sd card (you might add a warning note about that part), I had it swapped over to local extension numbers and trunked to existing asterisk servers in minutes… [I]t doesn’t get any cooler than this! — Scott P.

Trust us when we say the performance of this tiny computer is nothing short of amazing. Can it do everything a $300 dual-core Atom PC can do? No. Can it do 90% of everything for someone whose requirements do not exceed a handful of simultaneous calls at a time but still wants a full-blown PBX for call routing, transcribed voicemails delivered by email, IVRs, music on hold, and text-to-speech and speech-to-text apps for a home, a SOHO office, a Little League team, or a dorm room? Absolutely. As Scott mentioned, configuration changes may take you a few seconds longer than would normally be the case with an Atom-based PC and a hard disk. But, hey, you can have this delivered to your front door in two days with Amazon Prime shipping for less than $2 more than the actual cost of the $35 computer. See the ad in the sidebar for ordering info. You’ll have a fully functional PBX up and running before you can break a summer sweat.

And today we’re pleased to introduce Incredible PBX 3.11.10 for the Raspberry Pi, a turnkey PBX featuring the latest releases of Asterisk® 11 and FreePBX® 2.11 for a near perfect telephony platform. Special thanks to the tens of thousands of pioneers that have given the first dozen iterations of this software a healthy workout over the past 18 months. We couldn’t have done it without you!

What’s New in Incredible PBX 3.11.10? Well, just about everything is either new or upgraded. For those of you already running a previous version, here’s a quick thumbnail of the 3.11.10 feature set. As in the past, we’ve tried to mimic as much of the previous build functionality as possible while providing new firmware support for the very latest Raspberry Pi boards. So you still get simple utilities to configure 1GHz overclocking and automatic expansion of the 3.11 image to run on any size SDHC card. You also get Asterisk’s latest long-term support release, Asterisk 11, which provides an incredibly stable VoIP platform. There also have been some major plumbing enhancements in FreePBX 2.11 to improve its stability and to enhance security. Although we can’t promise it will last forever, you also still get Google Voice support with free calling in the U.S. and Canada as well as free faxing and SMS messaging plus most of the Incredible PBX feature set. We’ve also added optional voice transcription and email delivery of MP3 voicemails to smooth the migration to unified messaging. If you want the first 15 seconds of each voicemail transcribed, then issue the following commands after logging into your server as root. Additional documentation is available here.

cd /usr/sbin
mv sendmailmp3 sendmailmp3.notrans
mv /root/sendmailmp3 sendmailmp3

Incredible PBX 3.5 and beyond added automatic detection and support of 512MB Raspberry Pi devices without touching anything. Beginning with version 3.7, we added an awesome fax server to Incredible PBX for those with a 512MB board. The complete tutorial is available here. update-my-pi in the /root folder of your server helps to safeguard your system by bringing it up to date with the latest fixes and enhancements. After running it the first time, it gets run automatically whenever you log in as root. For the safety of your server, don’t disable it! It’s free for the first ten updates, and then it’s just $20 a year. And it’s on the honor system so you can cheap out if you prefer not to contribute.

To enable overclocking at your own risk, run: raspi-config. Overclocking works for us. YMMV! The key is a good power supply. If you’re using an SD card larger than 4GB, version 3.11.10 can resize your partitions on the fly. Just choose the option in raspi-config and reboot. Even though the kernel now monitors CPU temperature and manages overclocking, it’s always nice to see for yourself. To monitor the CPU temperature, just run the status program which provides a current snapshot anytime. Temperature data now is provided on the FreePBX Dashboard as well.

There’s more good news. Networking is much more stable; however, we’ve dropped support for the TP-Link WiFi adapters. If you still have one of these adapters, see this thread on the PIAF Forum for setup instructions. Otherwise you’ll need the recommended AirLink 101 N-150 if you want WiFi capability. In the 3.11.10 release, your server will automatically attempt to connect to any open WiFi network that it can find. We’ve also added two scripts in /root to let you restart either your wired or wireless network and designate it as the primary network: restart-eth0 and restart-wlan0. Time zone management was also a bit of mess with multiple file settings required to support both Linux and PHP. In this release, you’ll be prompted to select your timezone when you first log in as root. The setup script will automatically apply your entry in all the right places. We’ve also replaced SendMail with Exim to simplify the process of using an SMTP mail gateway such as Gmail. The procedure for making the change is documented here.

As part of the latest build, we’ve also eliminated the pi user account. Everything you need to do to configure Incredible PBX requires root permission. So goodbye sudo. Only the root user account is included, and the default password is raspberry. Change the password when you first log in. New SSH and DUNDI keys now are automatically generated when you first boot your server. The FreePBX Backup and Restore Utility is included in this new build. AsteriDex Speed Dialing has been enhanced for our friends across the Atlantic. Enter 3-digit Dial Codes in AsteriDex, and you can call by dialing 000nnn. In this upgrade, Telephone Reminders work like a champ.

X Windows is included in this build. This won’t work with SSH. For a demo slideshow, plug in a real monitor and log in as root. In the /root folder, enter the command: startx. To end the slideshow, press ESCape. To disable the slideshow: mv .xinitrc xinitrc. To add photos, copy .jpg images into /root/slideshow. No mouse is required for the slideshow but, if you run X Windows natively, you’ll need a USB mouse. The SMS Blasting app in /root now supports phone numbers (which use SMS) and email addresses (which use SendMail).

Last but not least, a sophisticated Conference Bridge has been added to Incredible PBX. If you route one of your inbound DIDs to the predefined IVR, users can press 0 and enter 1234 for the conference PIN to join the conference. Local extensions simply dial C-O-N-F. We’ve already tested a 9-person conference call with excellent results. But don’t take our word for it. Try it for yourself. Just call our demo Raspberry Pi AutoAttendant and take the Conference Bridge and a handful of other Incredible PBX™ apps for a test drive:

And here’s what the conference call looked like in the FreePBX Dashboard:

So you don’t have to jump around between articles, we’ve put together this Quick Start Guide that tells you everything you need to know to get up and running in about an hour. Most of that time will be consumed copying the Incredible Pi image to an SD card. So there’s plenty of time for lunch during the hour. Once your system is running and you’ve completed the setup steps below, then jump over to the application tutorial which explains how to use every one of the 35+ Incredible PBX Apps for the $35 Raspberry Pi.

What to Buy. Here’s everything you need to get started.¹ The case is optional. Yes, you can run the Raspberry Pi sitting on your desk with no case. It’s only 5 volts. WiFi is also optional. There’s a 10/100 port on the Raspberry Pi that gives you all the networking you need. Here are the links to buy the pieces. You also need a CAT5 cable and either a spare PC or Mac with Putty or SSH and a pair of earbuds or an HDMI cable to connect to a TV or monitor and a USB keyboard.

$35 512MB Raspberry Pi from MCM or Newark or Amazon
$10 Power Adapter (this one is rock solid!)
$10 SanDisk 16GB SDHC Type 4 card
$15 Adafruit Acrylic Case
$13 AirLink 101 N150 WiFi Adapter or Adafruit Mini WiFi Adapter

Setting Up Google Voice. If you want free calling in the U.S. and Canada, then you’ll need a Google Voice account, and you’ll need one dedicated to Incredible Pi, or it won’t work. Log out after setting up the new Google Voice account! Also note that Google Voice will cease to function on May 15, 2014. You can read all about it here.

Register for Google Voice account (no funky characters in your password!)
Enable Google Chat as Phone Destination
Configure Google Voice Calls Settings:

Call Screening – OFF
Call Presentation – OFF
Caller ID (In) – Display Caller’s Number
Caller ID (Out) – Don’t Change Anything
Do Not Disturb – OFF
Call Options (Enable Recording) – OFF
Global Spam Filtering – ON

Place test call in and out using GMail Call Phone

Baking Your Incredible Pi. The disk drive for the Raspberry Pi is an SD card. So what you need to do is download Incredible Pi and copy the image onto an SDHC card. Mac and Linux installers are included. For Windows, just use Win32 Disk Imager. Here are the steps:

Download the latest Incredible Pi package
Decompress the tarball: tar zxvf incrediblepi-version.tar.gz
Copy .img file to SDHC using image utility or script
Insert SDHC card into Raspberry Pi
Boot the Raspberry Pi from Incredible Pi SDHC card

Your First Bite of Incredible Pi. If you’re not using a monitor and keyboard, you can use SSH to gain root access to Incredible Pi. And you can use any web browser on your private network to access your server. There are a couple of hurdles. First, you need the network address of your new server. And, second you need an SSH client. With Incredible PBX 3.3 and beyond, you needn’t worry about the IP address. You now can access your server via SSH by logging in like this: root@incrediblepbx.local. And browser access to your server is available at the following address: http://incrediblepbx.local. You still can plug in some earbuds when the bootup process begins and listen for the Incredible Pi to tell you its IP address when the boot procedure completes (about 90 seconds). Then you can use that IP address instead of incrediblepbx.local. The latest releases of Incredible PBX also include a Java-based SSH client in the FreePBX web GUI: Admin -> Java SSH. Because of the almost weekly security problems with Java, we strongly recommend using a standalone SSH client such as Putty.

Here’s everything you need to know about security for Incredible Pi:

1. ALWAYS RUN INCREDIBLE PI BEHIND A SECURE HARDWARE-BASED FIREWALL/ROUTER
2. NEVER EXPOSE ANY INCREDIBLE PI PORTS DIRECTLY TO THE INTERNET
3. NEVER MAP INBOUND INTERNET PORTS FROM YOUR FIREWALL TO INCREDIBLE PI

Initial Setup. There also are a few setup steps to complete once your Incredible Pi finishes the bootup process. When you first login (username: root password: raspberry), you’ll be prompted to change your root password and to set your default time zone. If you’re using either a wired network or an open WiFi network, then everything just works. If you’re using secured WiFi, then you’ll need to plug in your credentials in /etc/wpa.conf and reboot. The wired network always takes precedence so unplug the cable if you want WiFi to be your primary network. /root/update-my-pi runs automatically when you log in as root. It will bring your server up to current specs. Finally, now’s the time to repartition your SD card if you’re using a card larger than 4GB (highly recommended!). While you’re at it, bump up the performance of your Raspberry Pi by 50% by setting the overclocking to turbo mode. It works great for us. YMMV! The key is a rock-solid power adapter such as the one we’ve recommended. Just run raspi-config and follow your nose.

Accessing Incredible Pi By GUI. You don’t have to be a Linux guru to use Incredible Pi. In fact, we’re just about finished with the Linux command prompt, but stay logged in until we finish the steps below. Most of your configuration of the PBX will be performed using the FreePBX® Web GUI.

If you’re new to Asterisk® and FreePBX, here’s the one paragraph primer on what needs to happen before you can make free calls with Google Voice. You’ll obviously need a free Google Voice account. This gets you a phone number for people to call you and a vehicle to place calls to plain old telephones throughout the U.S. and Canada at no cost. You’ll also need a softphone or SIP phone to actually place and receive calls. YATE makes a free softphone for PCs, Macs, and Linux machines so download your favorite and install it on your desktop. Phones connect to extensions in FreePBX to work with Incredible Pi. Extensions talk to trunks (like Google Voice) to make and receive calls. FreePBX uses outbound routes to direct outgoing calls from extensions to trunks, and FreePBX uses inbound routes to route incoming calls from trunks to extensions to make your phones ring. In a nutshell, that’s how a PBX works. There are lots of bells and whistles that you can explore down the road.

Let’s get started. Using a browser, enter the IP address of your server or just use incrediblepbx.local as the address. Choose FreePBX Administration. When prompted for a username and password, use admin for both. Here are the six steps you need to complete before making your first free call:

Change FreePBX admin Password and Default Email
Add Google Voice Account credentials
Set Destination for Incoming Calls
Change Extension 701 Passwords
Eliminate Audio and DTMF Problems
Install and Register a Softphone to Extension 701

1. Changing FreePBX admin Password and Default Email. From the main FreePBX GUI, click Admin => Administrators. Click on admin user in the far-right column. Enter a new Password and click Submit Changes button. Then click the Apply Config button. Next, set your default email address in the right margin of Admin -> Module Admin and save your entry.

2. Activating a Google Voice Trunk. To create a Trunk in FreePBX to handle calls to and from Google Voice, you’ll need three pieces of information from the Google Voice account you set up above: the 10-digit Google Voice phone number, your Google Voice account name, and your Google Voice password. Choose Connectivity -> Google Voice (Motif) from the FreePBX GUI. The following form will appear:

Fill in the blanks with your information and check only the top 2 boxes. If your Google Voice account name ends in @gmail.com, leave that out. Otherwise, include the full email address. Then click Submit Changes and Apply Config.

There’s one more step or your Google Voice account won’t work reliably with Incredible Pi! From the Linux command prompt while logged into your server as root, restart Asterisk: amportal restart

3. Setting a Destination for Incoming Calls. Now that you’ve created your Google Voice Trunk, we need to tell FreePBX how to process inbound calls when someone dials your Google Voice number. There are any number of choices. You could simply ring an extension. Or you could ring multiple extensions by first creating a Ring Group which is just a list of extension numbers. Or you could direct incoming calls to an Interactive Voice Response (IVR) system (we’ve actually set one up for you to play with).

By default, Incredible Pi is configured to route all incoming calls to a demo IVR that shows off some of the applications that come with Incredible Pi. You can change whenever you like by choosing Connectivity -> Inbound Routes -> Default. In the Set Destination section of the form, change the target to Extensions and then select 701 from the list. Then click Submit and Apply Config.

4. Changing Extension Passwords. From the main FreePBX GUI, choose Applications -> Extensions. Then click on 701 in the Extension List on the right side of your display. You’ll see a form that looks like this:

For now, we only need to make a few changes. First, you need a very secure password for both the extension itself and your voicemail account for this extension. The extension secret needs to be a combination of letters and numbers. The Voicemail Password needs to be all numbers, preferably six or more. Replace the existing 1234secret and 1234 with your own (very secure) entries. You also need to lock down this extension so that it is only accessible from devices on your private LAN. You do that with the deny and permit entries which currently are filled with zeroes. Leave the deny entry the way it is which tells Incredible Pi to block everybody except those allowed in the permit entry below. For the permit, we need the first three octets of your private LAN address, e.g. if your LAN is 192.168.0.something then the permit entry will be 192.168.0.0/255.255.255.0.

Finally, you need to plug in your actual email address in the Voicemail section so that voicemails can be delivered to you when someone leaves a message. You can also include a pager email address if you want a text message alert with incoming voicemails. If you want the voicemails to automatically be deleted from the server after they are emailed to you (a good idea considering the disk storage limitations of an SDHC card), change the Delete Voicemail option from No to Yes. That’s it. Now save your settings by clicking the Submit button. Then reload the dialplan by clicking on the red prompt when it appears.

In case you’re curious, unless you’ve chosen to automatically delete voicemails after emailing them, you can retrieve your voicemails by dialing *98701 from any extension on your phone system. You’ll be prompted to enter the voicemail password you set up. In addition to managing your voicemails, you’ll also be given the opportunity to either return the call to the number of the person that called or to transfer the voicemail to another extension’s voicemail box. And you can always leave a voicemail for someone by dialing their extension number preceded by an asterisk, e.g. *701 would let someone leave you a voicemail without actually calling you.

5. Eliminating Audio and DTMF Problems. You can avoid one-way audio on calls and touchtones that don’t work with these simple settings in FreePBX: Settings -> Asterisk SIP Settings. Just plug in your public IP address and your private IP subnet. Then set ULAW as the only Audio Codec.

6. Setting Up a Desktop Softphone. Incredible Pi supports all kinds of telephones, but we’ll start with the easy (free) one today. You can move on to "real phones" once you’re smitten with the VoIP bug. For today, you’ll need to download a softphone to your desktop PC or Mac.

7. Activating Voice Recognition for Incredible Pi Apps. Google now requires an API key to use their voice recognition services. This affects a number of Incredible Pi applications including the Yahoo Weather report service (951) which lets you say the name of a city to retrieve its weather forecast, dial by name to call anyone in your AsteriDex database (411), and the Wolfram Alpha almanac service which lets you look up almost anything (4747). The 5-minute tutorial to activate voice recognition is available in the PIAF Forum.

As we mentioned, the easiest way to get started with Incredible Pi is to set up a YATE softphone on your Desktop computer. Versions are available at no cost for Macs, PCs, and Linux machines. Just download the appropriate one and install it from this link. Once installed, it’s a simple matter to plug in your extension 701 credentials and start making calls. Run the application and choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of Incredible Pi, 701 for your account name, and whatever password you created for the extension. Click OK.

Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place your first call. It’s that easy!

Monitoring Call Progress with Asterisk. That about covers the basics. We’ll leave you with a tip on how to monitor what’s happening with your PBX. There are several good tools within the FreePBX GUI. You’ll find them under the Reports tab. In addition, Asterisk has its own Command Line Interface (CLI) that is accessible from the Linux command prompt. Just execute the following command while logged in as root: asterisk -rvvvvvvvvvv.

Activating SAMBA for Windows Networking. SAMBA is included for transparent access using the Windows Networking Protocol from PCs, Macs, and other Linux machines. As delivered, SAMBA is deactivated. For obvious reasons, we recommend you never activate root login access to SAMBA without a very secure password. If you wish to enable SAMBA on your server, here are the steps while logged in as root:

1. Set SAMBA password for user root: smbpasswd -a root
2. Change Windows workgroup from WORKGROUP, if needed: nano -w /etc/samba/smb.conf
3. Manually start SAMBA from command prompt: service samba start
4. If desired, set SAMBA to start on boot: rcconf and activate SAMBA option

Activating the PPTP VPN Client. If you’ve followed the Nerd Vittles tutorial and previously set up a PPTP VPN Server for your devices, then it’s pretty simple to add Incredible Pi to the mix by activating its PPTP VPN client. You’ll need the FQDN or public IP address of your VPN server as well as a username and password for VPN access to your VPN server. Once you have those in hand, log into Incredible Pi as root.

Lest we forget to mention, you cannot log into your PPTP server from an IP address on the same private LAN so you’d only use the PPTP VPN when your Incredible Pi is at a remote location.

Edit the connection template: nano -w /etc/ppp/peers/my-pptp-server. Insert the following text and replace myfqdn.org with the FQDN of your PPTP server, replace myname with your PPTP username, and replace mypassword with your PPTP password. Then save the file: Ctrl-X, Y, then Enter.

To test it, issue the following command: /etc/init.d/pptp start. When you run ifconfig, you should now see a ppp0 entry:

ppp0 Link encap:Point-to-Point Protocol UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1 RX packets:0 errors:0 dropped:0 overruns:0 frame:0 TX packets:0 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:0 (0.0 B) TX bytes:0 (0.0 B)
Remember, it won’t show an IP address if the Raspberry Pi and your PPTP VPN Server are on the same subnet (like ours). Once you install your Raspberry Pi in a remote location, you now can access it at the first IP address in your reserved PPTP IP address pool.

To permanently activate the PPTP VPN client on your Incredible Pi server, run rcconf. Scroll to the bottom of the list and highlight pptp. Press the space bar to select it for automatic startup when you boot your server. Then tab to OK and press Enter.

Activating Incredible Fax. With a 512MB Raspberry Pi, here are the 5 Simple Steps to activate Incredible Fax. The original tutorial is available here.

Download and Install Incredible PBX 3.11
Run the /root/fax-enable Script to Automatically Configure HylaFax
Using FreePBX, Add Additional, Dedicated DID and Inbound Route to Handle Incoming Faxes
Install Any Desktop HylaFax Client to Send Faxes via Print-to-Fax using any PDF
Reboot Your Server and Enjoy

Incredible Trunks. When you’re ready to try some other SIP providers, here is a quick Cheat Sheet courtesy of Kristian Hare, who translated our original setups into a spreadsheet. Just click on the image below to open it in a new window. Then click on the redisplayed image to enlarge it.

Configuring CallerID Superfecta. In order to match names with phone numbers, Incredible PBX includes a FreePBX application named CallerID Superfecta. Out of the box, Incredible PBX 3.11 will work fine if you remember to activate CallerID Superfecta whenever you create a new Inbound Route. The CNAM entries also will be displayed in your CDR reports. For those not in the United States, you may prefer to use a lookup source for your numbers other than the ones preconfigured in CallerID Superfecta. You will find all of the available modules on the POSSA GitHub site. Just download the ones desired into /var/www/html/admin/superfecta/sources and then activate the desired sources in Admin -> CID Superfecta -> Default. You can test your results and the performance using the Debug facility that’s built into the module.

Shutting Down Your Server. Last but not least, never just pull the plug when you want to shut down your server, or you may end up with corrupted MySQL databases. Then nothing will run. Instead, log into your server as root, and issue the following command: shutdown -h now. Enjoy!

Where To Go Next. Once you’ve done a little exploring, take a few minutes to read the complete tutorial on all 35 Incredible PBX applications for Raspberry Pi. A few require a bit of configuration before you start using them. And then you’ll want to explore Interconnecting Asterisk Servers with Incredible PBX and the Raspberry Pi. Enjoy!

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.

Originally published: Monday, June 23, 2014

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! Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won’t have to wait long for an answer to your question.

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. [↩]

Another Feather in the VoIP Cap: Fedora 20 Comes to PBX in a Flash

Monday, June 16, 2014 / 1 Comment on Another Feather in the VoIP Cap: Fedora 20 Comes to PBX in a Flash

Summer has always been the season for experimentation at Nerd Vittles, and 2014 is shaping up to be a banner year. Red Hat® has been a pioneer in all things Linux® so we were anxious to take their latest technologies for a spin. Nothing sums up Fedora™ 20 better than Cubieboard’s graphic (shown above). For those new to open source, Fedora is where the Bleeding Edge action is. If you like roller coasters and fast cars, then you’ll feel right at home with Fedora 20. Suffice it to say it’s the 10th year anniversary edition and the twentieth release of Fedora. What’s new? Well, almost everything. At the desktop level, you’ll be struck by how quickly Fedora components are closing the gap on Windows® and Mac OS X®. In particular, the new LibreOffice™ office suite will leave you wondering whether your favorite word processor has been ported to Linux. It’s that good. And graphics support in Firefox® is every bit as good as any browser you’re currently using on any platform. Of particular interest to us was Fedora’s new support for the Cloud and more important to VoIP, the ARM® platform. This opens up exciting possibilities with the Raspberry Pi® and especially the third generation BeagleBone Black® with its new 4GB flash drive. Is that enough trademark symbols for you? If we missed any owned Word®, our apologies. Yes, you all own your brand names. And no, we didn’t invent any of this. We’re in the aggregation business, trying to make all the pieces work together. For a look at everything new in Fedora 20, be sure to read Danny Steiben’s terrific review. While there still are a few rough edges, it’s actually a much better product than the graphic above might lead you to believe. Here is sample displaying a favorite webcam site, ours.

Our back yard as shown live on RemleysPoint.org

What does all of this have to do with VoIP? Everything. Fedora is where RedHat experiments with new technologies that ultimately find their way into Enterprise Linux® releases.¹ With last week’s release of RHEL 7, CentOS™ 7 cannot be far behind. And much of the new Linux technology found in Fedora will be coming to a VoIP desktop or Cloud near you very soon. So it was important for us to see just how well Asterisk® and FreePBX® perform using the current PBX in a Flash™ installer. If you’re one who likes to read the last page of a book first, we’d give Fedora 20 an A-. And we don’t give out many A’s.

Our tasks for today are three-fold. We’ll show you how to install PBX in a Flash on top of an existing Fedora 20 installation. Then we’ll show you how to roll your own Fedora Remix, a generic operating system that you can embellish and redistribute to your heart’s content (pursuant to GPL2) without worrying about RedHat’s legal beagles. Finally, we’ll provide a Fedora Remix appliance for VirtualBox® that will let you deploy and play with Fedora 20 and PBX in a Flash on your favorite desktop computer. The complete appliance setup takes less than 5 minutes on almost any Windows, Mac, or Linux desktop.

Installing PBX in a Flash Atop Fedora 20 or Fedora Remix

Before you can install the latest PBX in a Flash aggregation, you’ll first need an operating system platform on which to run it. In the case of Fedora 20, that means downloading and installing the Live Media Desktop Edition. For today, we’ll assume you’re installing Fedora 20 on VirtualBox, but any relatively recent desktop computer should work equally well.

You actually have two choices for your operating system: the Fedora 20 platform described above or the PIAF-FC20 Remix which is a superset of that platform. The choice is completely yours. The Fedora Remix is not provided or supported by the Fedora Project. It has been created and is maintained by Ward Mundy & Associates LLC on behalf of the PBX in a Flash Development Team. Download the official Fedora software from here. Or download the PIAF-FC20 Remix ISO from SourceForge.

Let us restate the obvious. This is Bleeding Edge technology. Only deploy it behind a hardware-based firewall with no Internet port exposure. It is not safe to deploy this aggregation on the open Internet. It’s your phone bill. 🙂

Create a new Fedora 20 Virtual Machine in VirtualBox:

Type: Linux Version: Fedora RAM: 1024MB Default Drive Options with 20GB+ space Create Settings->System: Enable IO APIC and Disable HW Clock (leave rest alone) Settings->Audio: Enable Settings->Network: Enable, Bridged Settings->Storage: Far right CD icon (choose your ISO) and click Live CD/DVD Start

Install and configure Fedora 20:

Start your Virtual Machine Start Fedora Live (be patient while it loads) Click: Install to Hard Drive Choose Language and Click Continue Click: Install Destination (do not change anything!) Click: Done Click: Continue (to autoconfigure disk) Click: Begin Installation Click: Root Password: password, password, Click Done twice Click: User Creation: admin, admin, password, password, Click Done twice Wait for Software Install and Setup to finish Click Quit Click Activities and Search for terminal Click Terminal icon shutdown -h now Close the VM window and choose Power Off Machine

Adjust Virtual Machine to Remove Live Image:

Settings->Storage: Click on Fedora Live ISO Click: (-) icon to remove Live ISO Confirm: Remove Click: OK Restart Virtual Machine

Fedora 20 Initial Setup:

Accept default kernel for boot Click: Admin user, enter password, Accept Choose Language, Keyboard Ignore Link to Data in Cloud, Next Start Using Fedora Click: X to close Gnome Help window Activities, Search: terminal Click: Terminal icon su root enter your password init 3 login again as root ln -sf /lib/systemd/system/multi-user.target /etc/systemd/system/default.target service sshd start chkconfig --level 2345 sshd on ifconfig (decipher IP address for SSH login) shutdown -h now

Once you login with admin:password, the Gnome Desktop will appear. Navigation is similar to a Windows or Mac desktop. Clicking on the Power button and then the Settings icon will bring up the Settings window. Clicking on Activities will bring up the Application list and Search bar. You can drag any of your favorite apps to the left toolbar for quick access. For our purposes, type terminal in the search bar to access a Terminal window. Switch to the root user: su root. And enter your root password. Then complete the steps above to configure Fedora 20 for PIAF installation. You will NOT lose your ability to also use Fedora 20 apps in graphics mode. Switch between modes with init 5 for graphics and init 3 for non-graphics, multi-user mode. PIAF installation and operation requires run level 3. But, once it’s installed and operating, you can easily switch to run level 5 to use desktop applications such as FireFox and LibreOffice. The advantages of a multi-purpose platform for those who travel or for deployment at small remote sites should be obvious. A single computer could be used to provide BOTH desktop computing services as well as a full-featured PBX with secure connectivity to home base.

Preliminary Steps for PIAF3 Installation:

Restart Virtual Machine Login as root using SSH so you can cut-and-paste cd /etc/yum.repos.d # change gpgcheck=0 for all repos yum -y install httpd* php wget nano iptables-services glibc.i686 sed -i 's|SELINUX=enforcing|SELINUX=disabled|' /etc/selinux/config systemctl enable httpd.service systemctl start httpd.service systemctl stop firewalld.service systemctl status firewalld.service chkconfig --level 2345 firewalld off cd /root wget http://pbxinaflash.com/piaf3-install.tar.gz tar zxvf piaf3-install.tar.gz ./piaf3-install

PIAF3 Installation Procedure:

Phase 1: Allow automated install to complete (2,000+ new components) Phase 2: Following reboot, choose option C to exit to command prompt Type: piafdl Flavor: PIAF-Green # expand the size of your SSH window now by doubling its size Enable make menuconfig option: Y Time Zone: your choice FreePBX: 2.11 Master password: your choice When menuconfig opens, press down arrow, right arrow, left arrow, and X Wait for reboot and login again Post-install script will run and leave you with a functional system

Congratulations! Enjoy your new Bleeding Edge VoIP platform.

Rolling Your Own Fedora Remix

We promised you a quick tutorial on building your own Fedora Remix using Fedora 20 as a base. It’s actually pretty easy and can be built using the platform you created above. After logging into your server as root, issue the following commands to create the ISO. When you’re finished, you’ll have the same Fedora Remix ISO that can be downloaded from our SourceForge site.

mkdir /root/remix cd /root/remix yum -y install livecd-tools spin-kickstarts wget http://pbxinaflash.com/piaf-fc20-remix-ks.tar.gz tar zxvf piaf-fc20-remix-ks.tar.gz livecd-creator --config=fedora-live-desktop.ks --fslabel=PIAF-FC20-Remix \ --cache=/var/cache/live

PIAF3 VirtualBox Appliance with Fedora Remix

You may be asking, "Why the knuckle drill with rolling your own remix when Fedora provides the ISO for you?" The short answer is because RedHat has rules (lots of them) on how you can redistribute their open source products (many of which aren’t theirs at all). Most of these rules address which trademarks of theirs can and cannot be used and under what circumstances. For anyone building virtual machines, it’s simply the cost of doing business with RedHat. If it were as easy as removing the fedora-logos, fedora-release, and fedora-release-notes packages and replacing them with the generic-logos, generic-release, and generic-release-notes packages as RedHat’s site suggests, life would be easy. Unfortunately, there are dozens of commingled dependencies that get broken by directly swapping out these RPMs on a live system. While it has never been legally tested, as things stand today, you are forced to build your own ISO with the appropriate packages in order to comply with Red Hat’s licensing rules. Once you’ve done that, then creation of virtual machine appliances with the remix operating system are straight-forward and simple to create. We have built a VirtualBox appliance that will provide you a functional system on any desktop computer in minutes. The .ova appliance (3.1GB) is available for download from SourceForge. For those that want to experiment with this exciting new platform without the installation hassle, this virtual machine appliance is for you. After downloading the .ova image, just follow our previous VirtualBox tutorial to get started. It only takes a couple of minutes. At a minimum, change your root password by running passwd and change your FreePBX maint password by running passwd-master.

What’s Still Broken with PIAF Running Fedora 20

The new PIAF install procedure lets us push the latest fixes to every system as necessary. These always get loaded the first time you log in as root and configure your network for DHCP access, and you can reapply the latest fixes by issuing the command touch /etc/firsttime and then logging out and back into your server as root. With the latest fixes, the bug list is tiny. Apache authentication for FreePBX now works just as it does in all other versions of PBX in a Flash. Simply log into the FreePBX web GUI with username maint and the maint password you set up with passwd-master. The original PIAF status application has been replaced with an interim status app that provides most of the same functionality as the original. Zend Guard loader remains broken because Fedora 20 uses PHP 5.5. It means FreePBX commercial apps will not yet function. With those two exceptions, PBX in a Flash running under Fedora 20 should be indistinguishable from PIAF running on other OS platforms.

We really need your help identifying bugs! Much of this platform will ultimately be part of the new CentOS 7 and RHEL 7 builds. If you happen to stumble upon problems, particularly in FreePBX which now is dependent upon a new version of PHP and the new MariaDB database engine which replaced MySQL, please post a comment on the PIAF Forum AND open a bug ticket in the FreePBX Issue Tracker. The PIAF and FreePBX Dev Teams take all bug reports seriously and appreciate your assistance. Enjoy!

Originally published: Monday, June 16, 2014

Need help with Asterisk? Visit the PBX in a Flash Forum.

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 for all of us.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

Did you happen to notice who is using someone else’s trademark in their brand name? But we digress. [↩]

Top 3 Asterisk Security Tips for 2014: WhiteLists, WhiteLists, and WhiteLists

Wednesday, May 21, 2014 / 2 Comments on Top 3 Asterisk Security Tips for 2014: WhiteLists, WhiteLists, and WhiteLists

We’ve devoted a lot of energy to Asterisk security over the years with our Primer on Avoiding the $100,000 Phone Bill and our 20 Failsafe Tips and our SIP Navigation Guide plus numerous tutorials on deployment of Virtual Private Networks to secure your servers and phones including NeoRouter, PPTP, and Easy OpenVPN among others. But, when it comes to ease of installation and use with rock-solid security, nothing comes close to deployment of WhiteLists with the IPtables Linux firewall that’s included at no cost with every major Linux distribution and with all of the Asterisk® aggregations including PBX in a Flash™ and Incredible PBX™. So we’re kicking off the summer with a careful look at the methodology behind IPtables and the Travelin’ Man™ tools developed to reduce the learning curve for new users.

Security, of course, is all about the "bundle of sticks." As we learned from Aesop’s Fables, the more sticks you bundle together, the more difficult it is to break the stick. We are by no means advocating that you drop all of the other tools at your disposal to improve the security of your Asterisk security. So, before we dive into WhiteLists, let’s spend a little time covering some of the other tools that are available and why those tools should not be relied upon exclusively.

1. Hardware-based Firewall. The PBX in a Flash project has cautioned users for years not to run Asterisk-based servers connected to the Internet without a hardware-based firewall between your server and the public Internet. Is it failsafe? No. Some hardware-based firewalls have been compromised either by the bad guys or by the NSA. Pardon the redundancy. The other problem with hardware-based firewalls is that they’re generally not available with cloud-based solutions. As the price of cloud computing has dropped and the cost and headaches of maintaining your own hardware has increased, more and more folks are considering cloud-based alternatives. Yes. Hardware-based firewalls should be deployed whenever possible. No. They won’t resolve all security concerns.

2. Fail2Ban. Once upon a time, a number of us thought that Fail2Ban was the answer to all security issues with Asterisk-based servers. In a nutshell, Fail2Ban scans your logs searching for failed attempts to log in to either SSH, FTP, Apache, SIP, or an email account. After a small number of failed attempts, Fail2Ban blocks further access from the IP address initiating the requests. There are two problems with Fail2Ban. First, software developers of the affected services continue to "improve" things with new and different error messages when login failures occur. Since Fail2Ban is searching for specific word matches to identify unsuccessful logins, the whole security mechanism fails when the "magic words" change unless everyone is extremely vigilant in maintaining the "magic word" lists AND updating the Fail2Ban rules on all of your servers. Our experience suggests that the bad guys find the new "magic words" long before everyone else which means there are gaping holes in Fail2Ban regularly. The other problem is supercomputers such as Amazon EC2 which makes enormous computing resources available to every Tom, Dick, and Harry. We’re mostly worried about the Dick that can hammer your little server every second with hundreds of thousands of attempts to crack your SIP or SSH passwords. The problem this poses is that most Linux servers never allocate a sufficient time slice to Fail2Ban to scan your Asterisk, Apache, and SendMail logs. Instead of blocking a bad guy after 3 failed login attempts, a bad guy using EC2 may be able to perform several hundred thousand login attempts before Fail2Ban ever detects a problem. Yes. Fail2Ban helps against the bad guy manually keying in passwords. No. Fail2Ban is all but worthless against a sophisticated denial of service attack on your server.

3. Virtual Private Networks. The beauty of virtual private networks (VPNs) is that all of your Internet traffic is encrypted and tunneled through private IP addresses that others can’t intercept. That was the theory until Edward Snowden came along and spoiled the NSA’s party. Yes. We’ve known that PPTP VPNs were vulnerable for a good long while. No. We didn’t know that the NSA (and presumably others) may have had the keys to your castle much longer… regardless of the VPN topology you may be using. The other problem with VPNs is that you need VPN connections for every device connecting to your server. Unfortunately, VPN technology is only available on a small number of SIP telephones, and the supported OpenVPN topology is one of the more difficult VPNs to deploy on a Linux server. Are VPNs better than nothing? Absolutely. Does a VPN provide failsafe communications security over the open Internet? Probably not.

4. Nothing Beats Secure Passwords. Amen. There was a time when some Asterisk-based servers were routinely set up with extension passwords of 1234 or the extension number itself. And outbound SIP trunks were deployed with no dialing rules. And administrators opened accounts with SIP providers with automatic credit card replenishment whenever the accounts ran out of money to cover calls. And no safeguards were put in place to restrict international calling. Little did these folks know that registering to a SIP extension on an Asterisk server provided a blank check for making unlimited calls to anywhere on the planet. Thus was born the $100,000 phone bill. Yes. Nothing Beats Secure Passwords for root, for SIP accounts, and for SIP and IAX trunks connected to commercial providers. But you also need to implement dialing rules for outbound calls that allow your callers to reach only the destinations desired, not the world. And your accounts with providers should always include limits and restrictions on international calls and should never include automatic credit card replenishment.

5. BlackLists. There was a time when blacklisting IP addresses was believed to be the ultimate solution to Internet security problems. Sounds great, doesn’t it? Just set up a database with the IP addresses of all the bad guys in the world, and all our problems will be solved. Problem #1: A new bad guy is born every minute. Problem #2: The bad guys learned how to use VPNs and other random IP address masquerading sites to disguise their true identity. Problem #3: Security vulnerabilities in many Windows-based machines allowed the bad guys to take control of these computers and do their dirty work from there. Problem #4: There are actually some good guys that live in Russia and China. Problem #5: The bad guys learned to poison the "bad guy list" to block essential services such as DNS, Google, Amazon, Netflix, Pandora, and your favorite bank and credit card companies. Yes. The theory of blacklists sounded great. No. Blacklists not only don’t work. They’re downright dangerous.

WhiteLists with IPtables: The Knight in Shining Armor

#165972620 / gettyimages.com

For the past few years, our Internet security focus has turned toward defining a methodology that works with all PBX in a Flash and Incredible PBX servers, whether they’re dedicated servers behind a hardware-based firewall or public on a cloud-based shared host. And the conclusion we’ve reached is that nothing beats the IPtables Linux firewall for rock-solid Internet security. The reason is its deep integration into the Linux kernel itself through Netfilter, "a set of hooks inside the Linux kernel that allows kernel modules to register callback functions with the network stack." Wikipedia provides an excellent overview for those with an interest. For our purposes, suffice it to say that IPtables examines inbound and outbound packets before any further processing occurs on your server. With our default setup, we typically allow all outbound traffic from your server. For inbound traffic, if the iptables rules permit access, the packet comes in for processing. If not, the packet dies at the door with no acknowledgement that it was even received. In laymen’s terms, if someone attempts to scan your server to determine whether web or SIP services are available, there will be no response at all unless packets from the scanning server’s IP address are permitted in the iptables rules configured on your server. You can determine which rules are in force with this command: iptables -nL.

The basic configuration and syntax of iptables rules can be daunting to those unfamiliar with the territory. And thus was born Travelin’ Man 3, our open source tool to simplify configuration of IPtables by allowing administrators to define WhiteList entries describing the types of services that were allowed access to a server from specified external IP addresses. The basic rules of the Travelin’ Man 3 setup for iptables are these: (1) outbound packets are unrestricted, (2) forwarded, established, and related packets are permitted, (3) inbound packets from the private LAN are unrestricted, but (4) inbound packets from the public Internet are dropped unless permitted by a specific iptables rule. Those rules include certain basic services such as time synchronization (TCP 123) as well as WhiteListed IP address entries for specific or generic services.

Installation is easy. Log into your PBX in a Flash as root and issue the following commands. NOTE: Travelin’ Man 3 is optionally available as part of Incredible PBX installs on the CentOS, Scientific Linux, and PIAF OS platforms. It is preinstalled on the Raspberry Pi and BeagleBone Black platforms with RasPBX. You can determine if it’s already installed on your server with this command: ls /root/secure-iptables. If the script exists, you’ve already got Travelin’ Man installed, but it may not be running so keep reading…

cd /root wget http://incrediblepbx.com/travelinman3.tar.gz tar zxvf travelinman3.tar.gz yum -y install bind-utils ./secure-iptables

Because PBX in a Flash and Incredible PBX servers are primarily designed to support telephony, Travelin’ Man 3 further simplifies the iptables setup by whitelisting the IP addresses of a number of the leading VoIP providers. These include Vitelity (outbound1.vitelity.net and inbound1.vitelity.net), Google Voice (talk.google.com), VoIP.ms (city.voip.ms), DIDforsale (209.216.2.211), CallCentric (callcentric.com), and also VoIPStreet.com (chi-out.voipstreet.com plus chi-in.voipstreet.com), Les.net (did.voip.les.net), Future-Nine, AxVoice (magnum.axvoice.com), SIP2SIP (proxy.sipthor.net), VoIPMyWay (sip.voipwelcome.com), Obivoice/Vestalink (sms.intelafone.com), Teliax, and IPkall. For the complete list: cat /etc/sysconfig/iptables (CentOS) or cat /etc/network/iptables (RasPBX).

The real beauty of Travelin’ Man 3 is you aren’t limited to our WhiteList. You can add your own entries easily using the TM3 scripts that are included in the /root directory. secure-iptables initializes your iptables setup and also lets you define a primary IP address or fully-qualified domain name (FQDN) that will always have access to your server. You must run this script at least once to activate IPtables on all platforms!

Once you have run secure-iptables, you can whitelist additional IP addresses by running add-ip. You can whitelist additional FQDNs by running add-fqdn. You can delete either IP addresses or FQDNs by running del-acct. As noted previously, you can check what’s authorized with the command: iptables -nL.

We’ve also included a custom script to restart IPtables gracefully: iptables-restart. The reason is because using the traditional restarting mechanism in IPtables will leave your server vulnerable (and IPtables inoperative) if a particular FQDN cannot be resolved. The iptables-restart script takes another approach and removes the offending rule from your whitelist, alerts you to the problem, and then restarts iptables without the offending entry. So all existing rules are put back in place and function as you would expect.

Finally, Travelin’ Man 3 includes a script that allows you to utilize FQDNs for users that may have ever-changing dynamic IP addresses. Steps #4, #5, and #6 in the original Travelin’ Man 3 tutorial will walk you through the Administrator set up which only takes a minute or two and never has to be touched again. Basically, a cron job script is employed to check for changes in the dynamic IP addresses you have identified with FQDNs. If changes are found, IPtables is restarted which updates the IP addresses accordingly.

Unfortunately, there was one group of end-users that weren’t covered by the Travelin’ Man 3 setup. This group included traveling salespeople or vacationing individuals that may land in a different city every night. Rather than relying upon an administrator to provide access to home base, these frequent travelers needed their own tool to manage their IP address as it changed. While this was supported through a web interface in Travelin’ Man 2, that setup exposed your web server to the public Internet and was burdensome for administrators to initially configure. Most importantly, it didn’t manage remote IP address access using IPtables which made coexistence with TM3 difficult. Thus was born Travelin’ Man 4.

Introducing Travelin’ Man 4: Managing WhiteList Access by Telephone

Travelin’ Man 4 is a new add-on for an existing Travelin’ Man 3 setup. It’s for those that wish to allow traveling individuals to manage their own whitelist access to PBX in a Flash or Incredible PBX using a telephone. An Administrator preconfigures accounts and passwords for the travelers together with the services to which they will have access on the server. Using any cellphone or hotel phone, the traveler simply dials a preconfigured number to access an IVR that will prompt the user for an account number and PIN. Unless you have a spare DID, you can grab a free one from IPkall.com to use with your Travelin’ Man 4 IVR. Once a user is successfully logged in, the IVR will prompt for the user’s IP address to be whitelisted on the server. Enter it using this format: 12*34*56*78.

Within a couple minutes, the new IP address will be properly formatted and then whitelisted in IPtables, and the traveler will be sent an email acknowledging that the account has been activated. Once the account is activated, the traveler can use a SIP softphone application such as Zoiper on any iPhone or Android phone or a softphone on any desktop computer to place and receive calls as well as to check voicemail on the remote PBX in a Flash server. For anyone that doesn’t know their current IP address, a quick visit to WhatIsMyIP.com will tell you. Travelin’ Man 4 is licensed under GPL2 so download a free copy. Then read the tutorial and give it a whirl. Enjoy!

Originally published: Wednesday, May 21, 2014

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

FMC: The Future of Telephony with Vitelity’s vMobile and Asterisk in the Cloud

Thursday, May 15, 2014

If making phone calls from a web browser is what you’ve always longed for, then you’re in good company with Google and its future direction in the telephony space. Call us old fashioned but this strikes us as a solution in desperate need of a problem. What’s wrong with a Plain Old Telephone or a smartphone for making connections with friends and business associates? The real head scratcher is the fact that the WebRTC and Hangouts push demonstrates that the wizards at Google are seriously out of touch with the next generation. Will our 14-year-old daughter use Skype or Hangouts or FaceTime? Sure. About once a month to chat with Grandma or to interact with cousins scattered around the country, it’s a terrific option. And the same is true in the business community. When you need to collaborate with a half dozen colleagues, conferencing applications are invaluable. But to meet 95% of day in and day out business requirements, a telephone or smartphone is the clear device of choice. So join us today in celebrating the end of Google Voice XMPP service and the beginning of a new and even more exciting VoIP era… sans Google.

#470660911 / gettyimages.com

Of course, if it were up to the next generation, telephone calls might completely disappear in favor of text messaging, Snapchat, Instagram, and any other platform that includes recorded photos or videos. Note the subtle difference. Kids really are not interested in live video interaction. They find posed images that tell a story much more appealing. Why? Because recorded photos and videos let users present their best face, their movie star pose, and their expression of what they want others to perceive they’re really like. In short, live video is too much like real life. Our conclusion for those targeting the next generation is you’d better come up with something better and quite different than Skype, Hangouts, and FaceTime.

It’s Fixed-Mobile Convergence, Stupid!

Now let’s return to our primary focus for today, the current business community. Suffice it to say, there are a dwindling number of what we used to call "desk jobs" where an employee arrives at his or her desk at 9 a.m. and leaves at 5 p.m. As more and more jobs are headed off shore, the telephone and smartphone have replaced the corporate desk as the most indispensable corporate fixture. Particularly in the American marketplace, what we see with most businesses is a management layer and an (upwardly) mobile force of salespeople, consultants, and implementers that interact primarily through PBXs in an office headquarters or home office together with smartphones for those that generally are on the road. Many of these Road Warriors don’t even have a home phone any longer.

#185230942 / gettyimages.com

The telephony Holy Grail for this new business model is Fixed-Mobile Convergence (FMC). It’s the ability to transparently move from place to place while retaining your corporate identity. Every employee from the night watchman in Miami to the salesperson making calls from a Starbucks in California to the CEO in New York has an extension on a PBX in the cloud together with the ability to accept and place calls using the company’s CallerID name and number, transfer calls, and participate in conference calls regardless of whether the phone instrument happens to be a desktop phone or a smartphone. Is this even possible? Well, as of last week, the answer is ABSOLUTELY.

Vitelity has been a long-time corporate sponsor of both the Nerd Vittles and PBX in a Flash open source projects so we were thrilled when we were offered a free, Samsung Galaxy S III to try out the new (live) vMobile service that took Best in Show honors at ITEXPO Miami in January. As Vitelity’s Chris Brown would probably tell you, it’s one thing to demonstrate a new technology at a trade show and quite another to bring it into production. But Vitelity did it:

What we want to stress up front is that we’ve received no special treatment in getting this to work. We received the phone, opened a support ticket to register the phone on Vitelity’s vMobile network, and plugged our new credentials into the phone so that it could be integrated into our PBX in a Flash server. Once the smartphone became an extension on our PBX, we could place calls through our PBX with the S3 using both WiFi and Sprint 3G/4G service. Switching between WiFi and cellular is totally transparent. The CallerID for all outbound calls was our standard PBX CallerID. We also could place calls to other extensions on the PBX by dialing a 4-digit extension while connected to WiFi or the Sprint network virtually anywhere. If you have 3-digit extensions, those are a problem over the Sprint network but we’ll show you a little trick to get them working as well.

Keep in mind that every call from the S3 goes out through the PBX just as if you were using a standard desktop phone as a hardwired extension. And it really doesn’t matter whether the S3 has a WiFi connection or a pure cellular connection on Sprint’s network. You receive calls on the S3 in much the same way. It’s just another extension on your PBX. If you want to add it to a ring group to process incoming calls, that works. If other users on your PBX wish to call the S3 directly using the extension number, that works as well. If you want to transfer a call, pressing ## on the S3 initiates the transfer just as if you were using a phone on your desk. When we say transparent convergence, we really do mean transparent. No recipient of a call from the vMobile S3 would have any idea whether you were sitting at a desk in the corporate headquarters in New York or in a seat on a Delta jet after landing in San Francisco. Both the call quality and the corporate CallerID would be identical. And your secretary on maternity leave at Grandma’s house still could reach you using her vMobile S3 by simply dialing your corporate extension.

So that’s the Fortune 500 view of the new VoIP universe. How about the little guy with a $15 a month PBX in a Flash server in the RentPBX cloud¹, a couple mobile sales people, and a handful of construction workers that build swimming pools for a living? It works identically. Each has an S3 connected as an extension on the PIAF cloud server. And calls can be managed in exactly the same way they would be handled if everyone were sitting side-by-side at desks in an office headquarters somewhere. The silver lining of cloud computing is that it serves as the Great Equalizer between SOHO businesses and Fortune 500 companies. Asterisk® paired with inexpensive cloud hosting services such as RentPBX lets you mimic the Big Boys for pennies on the dollar. We think Vitelity has hit a bases loaded, home run with vMobile.

vMobile Pricing

We know what you’re thinking. "Since you got yours for free, what does it really cost??" The Galaxy S3 (or S4) is proprietary running Trebuchet 1.0, a (rooted) CyanogenMod version of Android’s KitKat. You can purchase these devices directly from the Vitelity Store. Currently, you can’t bring your own device. The refurbished S3 is $189 including warranty. Works perfectly! That’s what we’re using. Next, you’ll need a vMobile account for each phone. Unless you’re a Nerd Vittles reader, it’s $9.95 per month. That gets you free WiFi calling and data usage anywhere you can find an available WiFi hotspot. And text messaging is free. For calls and data using Sprint’s nationwide network, the calls are 2¢ a minute and the data is 2¢ per megabyte ($20 per gigabyte). For us, a typical day of data usage with an email account and light web use costs about a quarter. YMMV! So long as you configure Android to download application updates when connected to WiFi, data usage should not be a problem unless you’re into photos and streaming video. Android includes excellent tools for monitoring and even curbing your data usage if this is a concern.

vMobile Gotchas

Before we walk you through the setup process, let’s cover the gotchas. The list is short. First, we don’t recommend connecting vMobile devices to a PBX sitting behind a NAT-based firewall, or you may end up with some calls missing audio. The reason is NAT and quirky residential routers. If you think about it, when your S3 is inside the firewall and connected to WiFi, it will have an IP address on your private LAN just like your Asterisk server. When your S3 is outside your firewall on either a cellular connection or someone else’s WiFi network, it will have an IP address that is not on your private LAN. Others may be smarter than we are, but we couldn’t figure a way to have connections work reliably in both scenarios using most residential routers. You can configure your S3’s PBX extension for NAT=No or NAT=yes, but you can’t tell Asterisk how to change it depending upon where you are. One simple solution is to deploy these phones with a VPN connection to your Asterisk server sitting behind a NAT-based firewall. The more reliable solution is to build your PBX in a Flash server in the cloud with no NAT-based firewall. Then use an IPtables WhiteList (aka Travelin’ Man 3) to protect your server. From there, you can either interconnect the cloud-based server with a second PBX behind your firewall, or you can dispense with the local PBX entirely. Either way will eliminate the NAT issues with missing audio. In both cases, use NAT=yes for the vMobile extension.

Another wrinkle involves text messaging. Traditional text messages work fine; however, MMS still is problematic unless you initiate the outbound MMS session with the other recipient. It’s probably worth noting that Google Voice never got MMS working at all despite years of promises. This wasn’t a deal breaker for us, but it’s a bug that still is being worked on.

Finally, there’s Sprint. You either love ’em or hate ’em. We really haven’t used Sprint service in about eight years. In the Charleston area, the barely 3G service still is just as lousy as it was eight years ago. But, if you live in an area with good Sprint coverage and performance, this shouldn’t be an issue for you. And vMobile works fine in Charleston. You just won’t be surfing the web very often unless you have hours to kill… waiting. Additionally, dialing numbers with less than 4 numbers is a non-starter with Sprint, but we’ll show you a simple workaround to reach 3-digit local extensions from your vMobile device below.

With a service as revolutionary as vMobile with Sprint’s new FMC architecture, we can’t help thinking there may be other cellular carriers with an interest in deploying this technology sooner rather than later. But, given the vMobile feature set, Sprint is good enough for now especially when WiFi connectivity is available almost everywhere.

vMobile Configuration at Vitelity

For the Vitelity side of the setup, you first configure your smartphone using the (included) My Phone app. When the application is run, your cellphone number will be shown. Tapping the display about a dozen times will cause the phone’s setup to be reconfigured. Vitelity will provide you the secret key to activate your account. Next, you’ll log into the Vitelity portal and choose vMobile -> My Devices under My Products and Services. The account for your vMobile device will already exist. Clicking on the pull-down menu beside your vMobile device will let you create your SIP account on Vitelity’s server. Enter the IP address or FQDN of your Asterisk server and set up a very secure password. Your username will be the 10-digit phone number assigned to your vMobile phone. Save your settings and then choose the Edit option to view your setup. The portal will display your Username, Password, and FreePBX/Asterisk Connect Host name. Write them down for use when you configure your new extension using FreePBX®.

vMobile Configuration for Asterisk and PBX in a Flash

On the PBX in a Flash server, use a browser to open FreePBX. Choose Applications -> Extensions and add a new generic SIP device. For Display Name and User Extension, enter the 10-digit phone number assigned to your vMobile device. Under Secret, enter the password you assigned in Vitelity’s vMobile portal. Click Submit and reload FreePBX when prompted. Then edit the extension you just created. Set NAT=yes and change the Host entry from dynamic to the FQDN entry that was shown in Vitelity’s vMobile portal, e.g. 7209876542.mobilet103.sipclient.org. Update your configuration and restart FreePBX once again. Finally, from the Linux command prompt, restart Asterisk: amportal restart. If you’re using a WhiteList with IPtables such as Travelin’ Man 3, be sure to add a new WhiteList entry for your vMobile Host entry. Finally, add your vMobile extension to any desired Inbound Routes to make certain your vMobile device rings when desired.

You now should be able to place and receive calls on your vMobile device. If you want to be able to call 3-digit Asterisk extensions on both WiFi and while roaming on the Sprint cellular network, then you’ll need to add a little dialplan code since Sprint reserves 3-digit numbers for emergency services and will reject other calls with numbers of less than 4 digits. Here’s the simple fix. Always dial 3-digit extensions with a leading 0, e.g. 0701 to reach extension 701. We’ll strip off the leading zero before routing the call. The dialplan code below works whether you’re calling a local 3-digit extension or a 3-digit extension on an interconnected remote Asterisk server. Simply edit extensions_custom.conf in /etc/asterisk and insert the following code at the top of the [from-internal-custom] context. Then restart Asterisk: amportal restart. Note that we’ve set this up so that, if you have an extension 701 on both the local server and a remote server, the call will be connected to the local 701 extension. If you have different extension prefixes for different branch offices (e.g. 7XX in Atlanta and 8XX in Dallas), then this dialplan code will route the calls properly assuming you’ve configured an outbound route with the appropriate dial pattern for each branch office.

exten => _0XXX,1,Answer exten => _0XXX,n,Wait(1) exten => _0XXX,n,Set(NUM2CALL=${CALLERID(dnid):1}) exten => _0XXX,n,Dial(sip/${NUM2CALL}) exten => _0XXX,n,Dial(local/${NUM2CALL}@from-internal) exten => _0XXX,n,Hangup

Vitelity vMobile Special for Nerd Vittles Readers

Now for the icing on the cake… We asked Vitelity if they would consider offering special pricing to Nerd Vittles readers and PBX in a Flash users. We’re pleased to report that Vitelity agreed. By using this special link when you sign up, the vMobile monthly fee will be $8.99 instead of $9.95. In addition, your first month is free with no activation fee. We told you last week that there was a very good reason for choosing Vitelity as your SIP provider. Now you know why.

And, if you’re new to Cloud Computing, take advantage of the RentPBX special for Nerd Vittles readers. $15 a month gets you your very own PBX in a Flash server in the Cloud. Just use this coupon code: PIAF2012. Enjoy!

Originally published: Thursday, May 15, 2014

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

RentPBX also is a corporate sponsor of the Nerd Vittles and PBX in a Flash projects. [↩]

Beware the Ides of May: It’s Hammertime for Google Voice

Thursday, May 1, 2014 / 8 Comments on Beware the Ides of May: It’s Hammertime for Google Voice

https://www.youtube.com/watch?v=NyEE0qpfeig

You don’t have to be a soothsayer to appreciate what’s about to happen in the VoIP community. In just two weeks, millions of telephones in the United States are about to go silent. Let’s begin with what we know and don’t know about Google Voice and Google’s May 15 deadline. Google has made it crystal clear that XMPP connectivity to Google Voice is going away on May 15. What that means is that inbound and outbound calling using an XMPP connection to Google Voice will no longer work, period. And the platform really doesn’t matter. That includes Asterisk, FreePBX, FreeSwitch, Yate, and GrooVe IP as well as hardware ATAs such has ObiHai devices. Why? The short answer is because Google says so, and they are/were paying the bills. The longer answer is that companies such as Microsoft and Apple that have proprietary communications platforms were not reciprocating with free connectivity to their services in the same way that Google was providing XMPP service. Another probable reason is that Google was taking a financial bath on Google Voice services which were being abused by many commercial organizations. Reportedly, as many as three to five million DIDs have been handed out as part of the Google Voice project with very little return on investment.

Some have suggested this is just another tempest in a teapot like Y2K. After all, Y2K came and went without many catastrophes. The difference is that businesses spent hundreds of millions of dollars preparing for Y2K to make certain there were no train wrecks. With Google Voice, many individuals have taken the ostrich approach with their heads buried in the sand pretending things are just going to work out. Without some effort on the part of those still using Google Voice, May 15 will be their Julius Caesar moment.

What to Do? One school of thought is that the "old fashioned" Google Voice connections using Python which simulated a web call with Google Chat will still function. If receiving and placing calls using your existing Google Voice numbers matters to you, take the opinions of these self-proclaimed experts with a grain of salt. Here’s what you need to appreciate. First, nobody outside of Google actually knows whether the Python approach will continue to function or not. Second, even if it works on May 15, nothing would preclude Google from making "adjustments" at any time that would disable this functionality. They’ve done it before. They can do it again. And Google has made it abundantly clear that they’re putting an end to the free gravy train. Third, it doesn’t take a rocket scientist to deduce that PSTN call forwarding using Google Voice may be the next axe to fall. This probably won’t happen on May 15, but who knows. Finally, should you decide to go down this road, be aware that it is a major coding project regardless of your platform. But, if this is the road you wish to travel, you can find some tips on making the transition here. You’ve been warned.

The Smarter Approach. Our recommendations today are limited to those in the United States. Our apologies, but that’s two-thirds of our readership and roughly 95% of those that currently rely upon Google Voice. The same recommendations apply to those in Europe and South America and the Far East if calls to destinations in the U.S. are a major part of your VoIP traffic. What do we recommend? First, become VoIP savvy! The provider you use for outbound calls need not be the provider you use for incoming calls. Not putting all your eggs in one basket is a very good idea in the VoIP world.

Call us Chicken Little if you must, but Outbound Calling with Google Voice is going away on May 15. So, in the next two weeks, you definitely need to come up with an alternative for call terminations in the U.S./Canada market. We think you have two options: purchase an all-you-can-eat plan that includes sufficient outbound calling minutes to meet your existing requirements. Or you can select a provider that offers pay by the minute service for all of your outbound calls. One advantage with most of the pay-by-the-minute providers is that you can set your CallerID as desired. Don’t be misled by the all-you-can-eat claims. Every VoIP provider imposes some sort of cap on outbound calling even if their plan is advertised as "unlimited." If your outbound calling minutes exceed 2000-3000 minutes a month, you’re going to be looking for a new provider within weeks because every provider that we know will drop you like a hot potato when you are no longer profitable in their business model. The other gotcha is that most, if not all, of the all-you-can-eat plans are restricted to residential (non-business) use.

Full Disclosure: We have a favorite all-you-can-eat provider (Vestalink) and a favorite pay-as-you-go provider (Vitelity), and both of them provide some financial support to the Nerd Vittles and PBX in a Flash projects; however, both were our favorites before they provided any support to our projects.

All-You-Can-Eat Calling Plans. We continue to like Vestalink (formerly Obivoice) even though their prices have increased since the release of our original article. That’s actually a good thing. There was no way they could have stayed in business with their original pricing model. On a new 2-year plan with unlimited U.S./Canada inbound AND outbound calls, E911 service, and a free DID in your choice of area codes, the current rate for 24 months is $89.99 which works out to roughly $3.50 a month. The service comes with a 30-day money-back guarantee.

Another option which we previously have covered is a hardware device such as the netTALK Duo. With an upfront $100 hardware investment, you get the same features as Vestalink for $30 a year which works out to less than 10¢ a day. With both services, you have the option of porting your existing Google Voice number for a one-time fee. With Vestalink, you also have the option of spoofing your outbound CallerID number with your existing Google Voice number once it is verified as belonging to you. We prefer the latter approach at least until Google gives some hint that their call forwarding of incoming Google Voice calls is going away. Both services are bargains in our view. But, as we noted, for residential service we still prefer the pure VoIP solution provided by Vestalink.

Pay-As-You-Go Call Terminations. Most of the reputable pay-by-the-minute providers charge between 1¢ and 2¢ a minute for outbound calls with charges billed in 6 to 10-second increments. Unless you make an enormous number of lengthy calls, these rates are a bargain. Vitelity remains our favorite provider primarily because of the flexibility their service offers in setting up multiple sub-accounts for use with Asterisk or FreeSwitch. A sign-up link with a 50% discount on most DIDs is provided here and at the end of this article. We appreciate your support of our VoIP projects!!

While it is not yet officially available, the most compelling reason to switch to Vitelity is vMobile, a new $9.99/month cellphone plan that will integrate your Vitelity cellphone (actually a Samsung Galaxy S III) directly into your Asterisk setup. What that means is calls to extensions on your Asterisk server will also ring on your cellphone. And your cellphone functions exactly like any other extension on your Asterisk server whether you’re operating on 3G, 4G, or LTE networks as well as on WiFi at your home or office. You’ll be able to park calls, transfer calls, set up call monitoring, conferencing, and recording just as if you were on a standard VoIP phone in your home or office. And you can’t beat the price. Inbound and outbound calls on WiFi are totally free. Calls received or placed over what appears to be Sprint’s nationwide network are 2¢ a minute, about the same cost as pure VoIP calls.

For pay-by-the-minute terminations, we always recommend you set up accounts with multiple providers. Then, by setting multiple trunk sequences in your outbound routes, you’ll always have successful calls even when a particular provider happens to have an outage. Other than perhaps a small deposit, redundancy costs you nothing since you only pay for calls that you actually place through each provider. For a current list of our favorite termination providers in both the U.S. and Canada, see this thread on the PIAF Forum.

Handling Incoming VoIP Calls. Here’s the bottom line. The one thing you don’t want to do is risk losing your phone number because of the Google Voice train wreck. We have noticed a dramatic difference in call reliability for incoming calls over the past few months. Perhaps it’s an upstream provider problem… and perhaps not. Whatever the reason, get your phone numbers ported out of Google Voice as quickly as you can. It doesn’t have to be in the next two weeks, but you are well advised to begin the porting process soon. The Nerd Vittles Vitelity link will get you a monthly rate of $3.95 for a Tier A DID with unlimited incoming calls each month and automatic server failover. There are a few less expensive DID providers but, when it comes to our phone number, we’ve always wanted a provider with rock-solid reliability, flexibility, and a proven track record. Vitelity meets those requirements in spades. As we noted at the outset, the other advantage in separating out your inbound and outbound trunks is that, when service gets disrupted (and it happens to the best of providers), you’re not completely dead in the water.

For the short term, so long as you have an existing DID in the U.S. or Canada, you can forward your incoming Google Voice calls to that DID by simply adding it as a call forwarding destination in your Google Voice profile. We also recommend adding your cellphone as an additional call forwarding destination. Finally, be sure to disable the Google Chat option in your Google Voice setup and remove the Google Voice trunk in your FreePBX Google Voice/Motif setup. Good luck!

Originally published: Thursday, May 1, 2014

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

Some Recent Nerd Vittles Articles of Interest…

The Perfect VoIP Sandbox or Is It? Meet Digital Ocean Cloud Hosting

Thursday, April 17, 2014 / 4 Comments on The Perfect VoIP Sandbox or Is It? Meet Digital Ocean Cloud Hosting

After a week of testing, we’re taking a second look at Digital Ocean, a ~~terrific~~ cloud platform for experimenting with VoIP. Our initial testing was performed during a week that many folks were on Spring Break. After Easter weekend, our own testing suggested a very different level of performance. That warranted a more sober reevaluation in our judgment. Before we get to the specifics, let us emphatically stress that we do not recommend Digital Ocean as a long-term, cloud-based VoIP solution because of some mixed reviews on overall service reliability. Go here for details. Our own testing confirms that performance can be problematic because of what looks to be server capacity issues. To their credit, Digital Ocean provides bleeding edge technology which is regularly updated. But, if rock-solid platform stability and reliable performance are your primary requirements, there is no finer service than RentPBX, a long-time supporter and contributor to the PBX in a Flash project. With the PIAF2012 coupon code, your first turnkey PBX in the cloud is still $15 a month with server locations throughout the world. The cost difference is negligible if the quality of your phone calls actually matter to you. RentPBX has a proven record as an extremely reliable provider. And, if your business depends upon reliable VoIP technology, RentPBX remains one of the best bargains on the planet.

For today, we’re exploring Digital Ocean which we believe offers a good platform for learning about Asterisk® and VoIP because of its flexibility in creating and deleting virtual machines in the cloud environment. As you can see from the screenshot above, the base Digital Ocean offering appears to be ideal as a cloud-based Asterisk server. The 512MB memory option with 20GB of SSD storage and a terabyte of monthly bandwidth costs $5 a month. You actually pay $0.007 an hour until you reach the $5 monthly cap. Unfortunately, depending upon server load, performance can be hit and miss.

Digital Ocean calls their images droplets, and you’ll have to pardon us for using the terms interchangeably. The bottom line is we were able to create a PBX in a Flash server with Incredible PBX 11 running atop CentOS 6.5 in less than an hour. And we played with it for another hour. See below for the total cost. Note that the meter continues to run with your droplets until you physically Destroy them from the Digital Ocean Control Panel. What our initial testing did not reveal was that at busy times of the day the droplet creation process can vary from almost instantaneous to a couple of hours.

Getting Started. Let’s walk through the entire process of creating a PBX in a Flash server and adding Incredible PBX 11 using a Digital Ocean droplet, and we’ll assume you hit a Digital Ocean server on a good day. First, you’ll need an account. You can sign up with our referral code and provide a little financial support to the Nerd Vittles project. That doesn’t cost you a dime. Here’s the link. As part of the sign up procedure, you’ll be prompted to enter a coupon code. SSDMAY10 will get you a $10 credit if you hurry. You still need to add at least $5 to your account either using a credit card or PayPal. We strongly recommend that you start with a minimal investment to make certain that Digital Ocean’s performance will meet your requirements. Test it regularly during your free trial period.

Once you’re registered, you can create your first Droplet by clicking on Droplets in the Digital Ocean Control Panel and then clicking the Create Droplet button. You’ll be prompted for a Hostname, the size Droplet you wish to create, the Region for your cloud-based server, and the Image Type for your server. We used PIAF512 for the hostname. The 512MB memory droplet will work just fine for experimentation. Choose a region that’s close to the provider you wish to use for VoIP calls. For the image type, choose a CentOS 6.5 32-bit server. In the Settings, leave Enable VirtIO checked and skip the Private Networking and Enable Backups options for the time being. Click Create Droplet to initiate the droplet build process which takes under a minute. Your new Droplet credentials will be emailed to you when the procedure is complete.

Installing PBX in a Flash. At this point, you have the option of logging into the Droplet from the Digital Ocean Control Panel, or you can use SSH to log in as root using the IP address provided in your Droplet creation email. Once you’ve logged in, issue the following command to set up your new server for the PIAF3 installer:
yum -y install wget nano

Now you’re ready to kick off the PBX in a Flash 3 installation. Here are the commands:
cd /root wget http://pbxinaflash.com/piaf3-install.tar.gz tar zxvf piaf3-install.tar.gz ./piaf3-install

The server will whir away for about 5 minutes (on a good day) configuring a CentOS superset of over 500 additional RPMs to support PBX in a Flash 3. When the setup is complete, your droplet will reboot. Count to 30 and then log back in as root and choose option A to install PBX in a Flash. Then choose PIAF from the menu of choices. Accept the license agreement and select PIAF-Green as your desired flavor. This gets you the latest Asterisk 11 release. When the Configuration Wizard loads, choose N to avoid loading Make MenuConfig for customization of Asterisk. Confirm your choice. At Step 2, choose your Time Zone and confirm your choice. At Step 3, choose FreePBX® 2.11 as your GUI and confirm your choice. At Step 4, create a master password for GUI and utility access. Tap Enter to confirm all of your selections one final time. Then take a coffee break. Depending upon server load, you’ll have a shiny new PBX in a Flash 3 server with CentOS 6.5, Asterisk 11, and FreePBX 2.11 in either 30 minutes (on a good day) or a couple of hours (on a not-so-good time of day).

Installing Incredible PBX 11. Once your server has rebooted following completion of the PIAF install, you’re ready to install Incredible PBX 11. This gets you dozens of preconfigured applications for Asterisk as well as a base configuration of FreePBX. You’ll still need to add a VoIP trunk to handle incoming and outgoing calls as well as configuring a phone to use your extension 701 credentials. Log back into your server and issue the following commands to install Incredible PBX 11:
wget http://incrediblepbx.com/incrediblepbx11.gz gunzip incrediblepbx11.gz chmod +x incrediblepbx11 ./incrediblepbx11

You can complete the configuration of your server using the FreePBX GUI. Using a browser, visit the IP address of your server. Click the Users button to display the PIAF Admin Control Panel. Then click the FreePBX option. Log in with username maint and the maint password you created above. Sign up for VoIP service with your favorite VoIP provider. Then create a trunk for that provider or edit and enable one of the preconfigured trunks that’s included in Incredible PBX. Download a softphone to your desktop and configure it using the credentials provided for the 701 extension in FreePBX. Then you’re ready to make your first call. See last week’s Nerd Vittles article for complete details.

Originally published: Thursday, April 17, 2014 Second Look: Wednesday, April 23, 2014

Need help with Asterisk? Visit the PBX in a Flash Forum.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.

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