Posts tagged: piaf

A Firsthand Look at Disaster Recovery: Tethering and IAX with Asterisk

One of the exciting challenges of building a swimming pool is knowing that it’s just a matter of time until your Internet connection dies. As you might imagine, swimming pools are major construction and involve a lot of digging. And digging usually means some oops moments when cables get cut. In our case, we had watched the folks digging the trenches for all of the pool plumbing to be sure they didn’t accidentally whack one of three coax cables coming into our house. And, when it came time to cover up the trenches, we pointed out the orange cables to the Bobcat driver knowing we were finally home free. Not so fast! Two minutes later, Mario had driven the Bobcat right over the primary Internet cable leaving the shredded remains sticking up through the dirt. Oops. Sorry. Shit happens!

Looking on the positive side, we chuckled, “What a perfect opportunity to test our backup Asterisk® system!” Our backup system is pretty clever if we do say so. It relies upon a Verizon WiFi HotSpot running on our Galaxy smartphone and a duplicate of our Asterisk-based PBX in a Flash™ server running as a virtual machine under VirtualBox on an iMac desktop. The entire setup takes less than a minute to activate. Well, that was the plan anyway.

It turns out that Verizon does SIP a little differently with a SIP ALG in the path so Asterisk couldn’t register with all but one of our dozen SIP providers. Congratulations, CallCentric! The workaround is to enable STUN. That is now possible with Asterisk 11. Short of that, you’re left with CallCentric. Unfortunately for us, we don’t do much SIP trunking with CallCentric, and none of our primary DIDs are connected through them. Our attention span also was too short to tackle STUN in the middle of this crisis. But there’s good news. Verizon doesn’t mess with IAX network traffic at all. And a couple of our primary DIDs are registered with VoIP.ms using IAX trunks. Restoring the IAX trunks to full functionality took less than a minute. That is step one of a three-step process. You need inbound trunks, phones, and outbound trunks to get your redundant VoIP server back in business.

Getting phones to function on what is now a purely WiFi network (through the Verizon HotSpot) can be problematic unless you’ve done your homework and sprinkled a few WiFi-capable SIP phones around your home or office. In our case, we still have Grandstream’s GXP2200 Android phones scattered everywhere so it was just a matter of plugging in the WiFI adapters and rebooting. The newer GXV3240 would work just as well.1

All that remained was enabling several trunks for outbound calls. Since VoIP.ms IAX trunks support both incoming and outgoing calls, we were home free. And, with Google Voice trunks, it was simply a matter of jumping through Google’s security hoops to reenable the connections on a new IP address.

Lessons Learned. Here’s a quick checklist for those of you that think about disaster recovery for your home or for clients and businesses. Nothing beats some advance planning. If money is no object, then WiFi tethering from a smartphone with one of the major providers whose service works well in your home or office environment is the way to go. 4G is a must!

In our case, money was an object so we had the foresight to acquire a Verizon SIM card from eBay that included an unlimited data plan. With this setup, it costs only $1 a day extra to add WiFi tethering, and you can turn it off and on as often as you like without any additional fees or surcharges. There also are no additional charges for using boatloads of data! We’re actually writing this column with a tethered connection from a hotel in Washington (results above). To give you some idea of why an unlimited data plan is important, our home operation burned through 4 gigs of data in less than 24 hours once we activated WiFi tethering. Of course, there were people doing things other than making phones calls, but tethering enables 5 connections to function just about like the cable modem service you originally had in place. So expect the data usage to be substantial. Everybody likes 24/7 Internet service.

Loss of phone calls through a PBX is more of an annoyance than a crisis these days because almost everyone also has a smartphone. Even so, the SIP gotcha with Verizon Wireless was a surprise because we hadn’t really tested our super-duper emergency system in advance. That wasn’t too smart obviously. The old adage applies. Do as we say, not as we do. Unplug your cable modem or DSL connection and actually test your backup system before D-Day arrives.

On the VoIP provider end, now is the time to set up an account with a provider that offers both SIP and IAX connectivity. Step 2 is to actually configure an IAX trunk (as a subaccount to use VoIP.ms parlance) and test it. IAX trunks actually have fewer headaches with NAT, but there are only a handful of providers that still provide the service. Find one now and make certain that your primary DIDs will roll over to the IAX trunk in case of an outage. I’m always reminded that we have Mark Spencer to thank for IAX. It was his brainchild. Thank you, Mark! With VoIP.ms, you also can spoof your CallerID so that calls will still appear to originate from your primary Asterisk PBX.

Keep in mind that a VirtualBox-based Asterisk virtual machine and a Desktop computer both need an IP address and will have to be started on WLAN0 rather than ETH0. Remember, your wired connection is now dead.

You’re also going to want to acquire at least a couple of WiFi-capable SIP phones that can be connected with your Asterisk server using your WiFi HotSpot. Also make certain that you have a preconfigured IPtables firewall on your backup system. Remember, your hardware-based firewall connected to your cable modem won’t provide any protection once you switch to HotSpot operation. Lucky for you, Incredible PBX™ servers come preconfigured with a locked-down IPtables firewall and a WhiteList. Just add the new IP addresses of your server and phones, and you’re secure on the public Internet.

Finally, let’s do the HotSpot connection math. You’ll need an IP address for your desktop computer running VirtualBox. You’ll need a second IP address for the Asterisk virtual machine. Then you’ll need an IP address for every WiFi-enabled SIP phone. If the maximum number of connections is five on your HotSpot, that means you’ve got the necessary capacity for at most 3 WiFi SIP phones assuming you don’t enable a WiFi printer and if nobody else wants to use a computer during the outage. The other option is to add an inexpensive travel router with bridge mode to your mix of 5 devices. We always keep one handy for extended trips. A properly configured travel router provides an additional WiFi network with some extra WiFi connections. Good luck!



Security Alerts. Serious SSL and FreePBX security vulnerabilities have been discovered AND patched during the past week. If you have not patched your server and Asterisk, FreePBX, Apache, and/or WebMin are exposed to the public Internet, you have a serious problem on your hands. See this thread for details on the FreePBX vulnerability. And see this thread for the steps necessary to patch SSL in Asterisk, Apache, and Webmin. While Incredible PBX servers were automatically patched for the FreePBX vulnerability, the SSL issues require manual patching and an Asterisk upgrade. A script for upgrading Asterisk 11 servers is included in the message thread linked above. ALWAYS run your VoIP server behind a firewall with no Internet port exposure to Asterisk, FreePBX, SSH, or the Apache and Webmin web servers! And, if you think all of this security stuff is just a silly waste of your time, then read about the latest lucky recipient of a $166,000 phone bill.

Originally published: Monday, October 20, 2014



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

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

Zero Day Vulnerability Protection and More: Introducing Cover Your Asterisk


It’s been a difficult couple of weeks for the Linux® and Asterisk® communities with the back-to-back disclosures of the BASH Shellshock bug and then the FreePBX® Asterisk Recording Interface (ARI) bug a few days later. Both of these vulnerabilities have been circulating in the wild for years. We won’t repeat Wikipedia’s Zero Day Attack analysis other than to note that what makes these particular bugs so scary is not only the fact that both went undetected and unpatched for years but also that the attack vectors for both bugs were so simple. Anyone with a web server exposed to the Internet that was running any flavor of Linux or any Asterisk server with the FreePBX GUI was fair game for a seriously compromised server.

For those with shared servers in a hosted environment running under cPanel, your web platform typically runs with the equivalent of root privileges which means that any web intrusion inherits the same server privileges that you as the administrator had. This is similar to the way FreePBX runs with Asterisk. The same user account used for web access controls all of the Asterisk assets on your server. While it’s convenient, it’s also dangerous whenever there’s a web vulnerability because the entire Asterisk platform has exposure.

We always chuckle when one of the anonymous forum trolls launches a tirade claiming that these alerts are nothing more than Monday morning quarterbacking disguised as Chicken Little. What’s more amazing is that anyone would take the comments of an anonymous poster seriously especially on a matter involving server security. It’s one thing to label folks as alarmists for suggesting that the sky is falling when it isn’t. It’s quite another to launch these anonymous personal attacks even when there is documented evidence that the Internet sky was indeed caving in. Kinda reminds us of the global warming naysayers when the polar ice caps are melting beneath their feet.

According to the naysayers, we’re all doomed when it comes to cyberterrorism so why fight it. Here’s why. While reacting to security vulnerabilities has always been a defensive game of cat and mouse, that doesn’t mean you shouldn’t proactively do what you can to patch serious security holes in your servers. The alternative is to give cybercriminals a blank check to launch bots from your server that generate spam or participate in large-scale zombie attacks on our most trusted resources whether they’re DNS root servers, utility infrastructure and our electric grid, banking assets, and even national security resources. So let’s circle back and address what you can do to assure that you’re part of the solution rather than part of the problem.

The Way It Is: Do I Need a Public Web Server with Asterisk?

For purposes of this discussion, our focus today is Asterisk server security. And the number one thing you can do to insulate your server from these vulnerabilities is to make certain that your web server is not exposed to Internet access by the general public. Neither Asterisk nor FreePBX requires public web server access to manage your server. In fact, neither Asterisk nor FreePBX requires any public access to your server to properly perform all required telecommunications functions. And the second paragraph above explains why this is especially dangerous with servers running both Asterisk and FreePBX.

So why do people still publicly expose their web servers and UDP ports 5060 and 10000-20000 to the Internet? As much as we hate to say it, it’s because it’s always been done that way. It’s also because there are a handful of SIP providers that still require UDP 5060 access to make and receive calls. Most do not! And even for those that do require UDP 5060 access, their requirements can be satisfied with a properly configured firewall that supports whitelisting of “safe” IP addresses for limited access. Incredible PBX comes preconfigured with a locked down WhiteList. The same can be added to PBX in a Flash by installing Travelin’ Man 3. We hope the other aggregations will follow suit. It’s long overdue.

Public web server access often is because there are more than a few (lazy) VoIP providers that install systems in a way that makes it easy for them to manage remote sites. Of course, a VPN would provide secure access to the same resources but that’s a little more work on the deployment end. With NeoRouter VPN, it’s a 5-minute job!

There also are companies with remote users or traveling salesmen that claim their servers must be open to the Internet to keep the company running. First, it’s hard to imagine a company whose salespeople don’t have cellphones that require no link to home base. Second, there are numerous solutions for safe connectivity with a home office: VPNs, FQDNs with dynamic DNS support, Port Knocker, and Travelin’ Man 4 to name just a few of the ones we previously have recommended. With the exception of the lazy VoIP installer, you will note that none of the above scenarios ever require web access to a VoIP server. So the rationale for public exposure of an Asterisk web server is all but non-existent.

The bottom line is that, if your server is not and has never been accessible from the Internet by typing its IP address into a public web browser and assuming your root password has not been compromised, then the BASH and ARI vulnerabilities are purely an academic discussion from your vantage point. Should you apply the patches anyway? Absolutely. Will your server be compromised if you don’t? Probably not… at least not from these two vulnerabilities.

Life Is Good: Why Do I Need ‘Cover Your Asterisk’

That brings us to our topic for today. Having said all of the above, how do you really know if your server has been compromised by some zero day attack vector that none of us yet know about? After all, there are tens of thousands of applications installed on a typical Linux server. And a zero day vulnerability could be hiding almost anywhere.

First, a few words about what Cover Your Asterisk is not. This application won’t detect previously compromised servers! Wearing a condom the day after your wild night on the town isn’t all that helpful. If your server has been running as a public web server for the last 5 years, then our best advice is to start with a fresh install to a new, secured server. Then manually copy the settings (not the files!) from your old server to the new platform. Now you’re ready to protect your server.

Second, more than a few words about the VoIP environment in which we find ourselves. If you’re running any of the so-called Asterisk aggregations including PBX in a Flash, Incredible PBX, AsteriskNOW, FreePBX Distro, or Elastix, then your server includes some flavor of the FreePBX GUI, a web-based application to manage and configure Asterisk. As part of the FreePBX GUI setup, you give FreePBX 2.11 and beyond an expansive set of privileges on your server. These include read, write, and delete access to all of your web assets, all of your VoIP-related MySQL database assets, and all of your Asterisk assets. You also grant FreePBX rights to inventory and monitor critical pieces of information about your server so that you can be informed about pertinent FreePBX updates. We don’t see this as a bad thing. But, even with the incredibly talented FreePBX development team, this application design can be dangerous for a number of reasons not the least of which is the events of the past week. Consider for a moment a scenario in which a disgruntled employee or a web vulnerability allows somebody to modify a critical Asterisk configuration file such as manager.conf which controls access to the Asterisk Manager Interface, or to adjust MySQL’s admin.ampusers table which controls web access to the FreePBX GUI, or even to insert a malicious module into FreePBX which “looks and feels” like part of FreePBX. When you don’t know what you’re looking for, detecting subtle changes can be extremely difficult even for the most talented people in the business. For everyone else, it’s next to impossible. This is especially true when the changes aren’t noticeable in the standard day-to-day operation of your server. That was what led us to conclude that an additional detection mechanism was essential to highlight hidden changes made to any of the critical components that make up the Asterisk platform. Thus was born Cover Your Asterisk.

The Elastix folks apparently weren’t comfortable with this arrangement and forked FreePBX years ago and moved to a self-managed environment. The drawback has been their pace of releasing updates and patches, and that apparently applies to the unaddressed ARI bug as well.

The remaining aggregations all function as we’ve described. Before we delve into Cover Your Asterisk, here’s a little known tip. On the output side, FreePBX is basically a code-generator for Asterisk. Once you’ve configured your server using the FreePBX GUI, there is no Asterisk-FreePBX linkage of which we’re aware that requires your web server to remain operational. That turns out to be a good thing. What this means is you can shut down Apache and still have a fully functional Asterisk server with all of the functionality of your FreePBX-designed configuration. Given the times in which we live, that may not be such a bad idea.

An Overview of Cover Your Asterisk

So what does Cover Your Asterisk do? What we’ve sought to do with this GPL2 application is to take a snapshot of your most valuable Asterisk and FreePBX assets and then create checksums of all the individual components. This includes the /etc/asterisk, /var/www/html/admin, and /var/lib/asterisk/agi-bin directories as well as the Asterisk DB and MySQL’s asterisk database. Periodically, you then run another script which compares your current setup to the previous snapshot and identifies the changes for further examination. Once you are satisfied that any reported changes are legitimate, you then take a new snapshot of your server and periodically check it to make certain no unexpected modifications have crept into your system. A duplicate of these production assets is always maintained in a separate directory structure (/etc/asterisk.snapshot) accessible only by root. It can easily be converted into a gzipped tarball: tar -cvzf cya.tar.gz /etc/asterisk.snapshot. Then simply store the tarball off site for a rainy day emergency… when the sky falls once again.

Because this application was designed for production servers, its testing and scope have been limited to the Asterisk 11 and FreePBX 2.11 platform. For our installed base, that translates into PIAF-Green with FreePBX 2.11 and all flavors of Incredible PBX 11 running atop CentOS, Scientific Linux, Ubuntu 14, Debian, and Raspbian platforms on both Intel and ARM hardware including the Raspberry Pi, BeagleBone Black, CuBox, and PogoPlug.

Installation and Operation of Cover Your Asterisk

Log into your Asterisk 11 server as root and issue the following commands to install the Cover Your Asterisk software:

cd /root
wget http://incrediblepbx.com/cover-your-Asterisk.tar.gz
tar zxvf cover-your-Asterisk.tar.gz
rm -f cover-your-Asterisk.tar.gz

To take the original snapshot of your server, run: /root/protect-your-ASSets.sh

To check your current setup against the snapshot, run: /root/check-your-ASSets.sh

To compare a file with its snapshot, run: diff /dirpath/filename /etc/asterisk.snapshot/dirpath/filename

To restore a snapshot file to your current Asterisk configuration, run these commands:

cp -p /etc/asterisk.snapshot/etc/asterisk/filename /etc/asterisk/filename
amportal restart

For Raspberry Pi and BeagleBone Black users, change the MySQL root password in both scripts:

sed -i 's|passw0rd|raspberry|' /root/protect-your-ASSets.sh
sed -i 's|passw0rd|raspberry|' /root/check-your-ASSets.sh

Finally, let us close with several recommendations. First, before making changes to your server with FreePBX, always run check-your-ASSets.sh, correct any detected problems, and then run protect-your-ASSets.sh to create a new snapshot of your server. After making any changes with the FreePBX GUI, run check-your-ASSets.sh again to verify that the changes you sought to make were, in fact, the changes that actually were made to your server. Then finish up by taking a new snapshot. These scripts take less than 30 seconds to run on a typical server so this is not a cumbersome process.

Before you restore any snapshot file or if you are puzzled by any changes you see listed after running check-your-ASSets.sh, we strongly recommend that you first seek advice from the gurus on the PIAF Forum. They can help you identify the severity of the problem, if any, and recommend an appropriate course of action for correction of the problem.

Finally, a cautionary note. Cover Your Asterisk is still a project in development. This means there will be changes/improvements as the coming weeks go by. One wrinkle with updates is your previous snapshots will have to be checked before you update. And then the newest protect-your-ASSets.sh script will need to be run following the update. To keep track of future releases and what’s included, visit this development thread on the PIAF Forum. Enjoy!

Originally published: Monday, October 6, 2014



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

Hold On to Your Wallet: Another Huge VoIP Phone Bill May Be Lurking


We interrupt our regularly scheduled content to bring you an urgent security alert. A couple days ago, a FreePBX® user reported unusual call activity. He traced the calls to a System Admin Dashboard module that was linked back to an IP address in the Netherlands. When the problem was reported, the FreePBX Community Manager quite accurately noted that it wasn’t FreePBX code. When a second user reported the exact same exploit, alarm bells apparently went off.

Further digging by the FreePBX Dev Team found that the legacy ARI module (once again) had been compromised, this time with a Remote Code Execution and Privilege Escalation exploit. Previous security vulnerabilities in this module led the PBX in a Flash developers many years ago to abandon the FreePBX security model in favor of Apache security so that we could totally block ARI access unless the user had administrator privileges. We want to stress that this wasn’t the fault of any of the current FreePBX developers. Instead, our move to Apache security was based upon our realization that this old legacy code was difficult to maintain because none of the original developers were still around. To their credit, the FreePBX developers have introduced a new User Control Panel with the strongest recommendation that the older ARI module be abandoned. Unfortunately, it still exists on all but the very latest FreePBX 12 systems including FreePBX 12 systems which were upgraded from a previous release. In addition, FreePBX 12 now provides checksum protection for all registered modules which will go a long way toward eliminating attacks such as this. So what can you do to protect your servers and your wallet today? For openers, upgrade your FreePBX fw_ari module NOW and clean the malicious module off your server:

rm -rf AMPWEBROOT/admin/modules/admindashboard
amportal a ma upgrade fw_ari

If you encounter an error that FreePBX cannot connect to the Asterisk Manager, do the following from the Linux CLI:

sed -i 's|localhost|127.0.0.1|' /etc/freepbx.conf
amportal restart
amportal a r

Protecting Your Server from Remote VoIP Attacks

Let’s approach the long-term solution on several levels starting with vulnerability exposure. If you can access TCP ports 22 (SSH) and 80 (HTTP) and TCP/UDP port 5060 (SIP) of any of your Asterisk® and FreePBX-based servers anonymously from the Internet, you’re either nuts or rich.

We’ve cautioned against this for nearly a decade and yet even some developers still configure Asterisk and FreePBX-based servers with port 80 Internet exposure. Why? We can only assume it’s because it makes their job of accessing and maintaining these systems easy. Don’t do it! There still are numerous ways to gain access to the FreePBX GUI on any server. Here’s our short list…

Safest. Put your server behind a hardware-based firewall with no Internet port exposure. Then use a VPN to access the FreePBX GUI. In a perfect world, you can run a VPN on all of your VoIP phones so that you have end-to-end protection for your server and all of your users.

Safer. If a hardware-based firewall isn’t possible, use the Linux IPtables firewall and lock down all the ports on your server, especially TCP ports 22 and 80 and TCP/UDP port 5060. Then create a WhiteList of IP addresses that need access privileges. It’s worth stressing that Fail2Ban is completely worthless when it comes to security vulnerabilities such as the ARI RCE flaw because the bad guys walk right in without even being challenged for a password.

Safe. If you need remote access from various remote locations and these sites have dynamic IP addresses, then deploy the Port Knocker technology in addition to locking down your server with the IPtables firewall. This lets you gain temporary access to your server without providing a blank check (literally) to everybody on the Internet. There’s a reason it’s called the World Wide Web and not the Good Guys Web!

Worse. Exposing TCP port 5060 and UDP port 5060 to public Internet access is dangerous. Some providers unfortunately still require direct access to 5060 to make VoIP calls with SIP. TIP: Switch to a provider that allows SIP registrations so that you don’t have to expose port 5060 directly to the Internet EVER!

Worser. Pardon our grammar, but exposing TCP port 22 to public Internet access is a bad idea. At the very least, change the SSH port so that typical port scanners don’t discover your open SSH port. SSH has been compromised in the past. It probably will happen again, or it may have already happened and we just don’t (yet) know about it. Fail2Ban helps with SSH attacks, but it’s not infallible particularly when high performance servers are used in the attacks. Fail2Ban has to scan your logs and, before it can do that, it has to have a sufficient time slice to accomplish the scan, something that may never happen with an attack launched from a platform such as Amazon EC2.

Worst. Never expose TCP port 80 to public Internet access. If you do, then you obviously haven’t had the pleasure of trying to maintain a public web server. TIP: Unless you are a web expert or sleep with one, don’t do it EVER! Earlier this week BASH provided a revolving door to your Internet assets using simple web requests. Earlier this year, OpenSSL was compromised. There will be another vulnerability because it’s the easiest attack target. So it’s just a matter of time until your server is compromised unless you deploy an effective firewall that blocks public access to port 80.

Server Design Still Matters

For our own PBX in a Flash and Incredible PBX users, you can sleep well tonight. Today’s vulnerability is mostly academic for you. PBX in a Flash blocks all access to ARI without the maint password. Incredible PBX blocks all access to ARI through its IPtables WhiteList. It’s still a good idea to apply the FreePBX update just to be double-safe. And Incredible PBX users will have the patch applied the next time they log into their server as root. For everyone else using FreePBX, keep reading.

With our Incredible PBX open source project, we provide state-of-the-art security methodology. While it is not infallible, all of the code is freely available for any and all VoIP developers to review, improve, and deploy. We would encourage our fellow VoIP developers to do so. There were reasons in the past for not deploying Apache security. After all, it lacks the flexibility of the FreePBX security model, and Apache also can be compromised. But we can’t think of any reason today for not deploying a hardened, preconfigured IPtables firewall AND a functional WhiteList as an integral component in every VoIP server install. This is especially important for any product deployed with the FreePBX GUI. Our Travelin’ Man 3 WhiteList implementation has been available for more than 2½ years! While there are downsides to any sort of push technology, we also believe the Incredible PBX (opt-in) update service is worth a careful look. It has been a godsend for us. With every new login, the server checks for important updates and processes them unless the administrator chooses not to use the service.

Keep in mind that FreePBX masquerading as the asterisk user has complete read/write privileges to virtually every Asterisk and web asset on your server. Any compromise is extremely dangerous because the asterisk user on these platforms has such expansive privileges. We recently encountered a trojan authorization lurking inside the permissions list of Asterisk’s manager.conf table. The matter is still under investigation so we can’t reveal much more other than to note that the entry was harmless on the few affected Incredible PBX servers because of the hardened IPtables WhiteList which is a key component of every Incredible PBX server. Had this happened on a server with no firewall protection, the intruder would have had complete access to the Asterisk AMI which pretty much gives the intruder a blank check to Asterisk… using your checkbook. The silver lining was the Incredible PBX update utility which provided a quick way to remove the vulnerability.

The FreePBX Dev Team’s efforts to design and deploy a checksum-based system for FreePBX 12 modules is certainly a step in the right direction. We think more safeguards are warranted. We already are exploring new ways to provide alerts when critical Asterisk or FreePBX resources are modified on PBX in a Flash and Incredible PBX servers. Something akin to the Mac’s admin authorization requirement before critical Asterisk or FreePBX changes are made would be ideal, but we have some other ideas as well. Stay tuned!

Originally published: Wednesday, October 1, 2014



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

VoIP Navigation Guide: Getting Started with Asterisk and FreePBX


When you were just getting started with Asterisk® in the early days, you had two choices: hire a consultant to build you an Asterisk system or start with Asterisk@Home and learn it yourself. That was a disaster for many folks. Times have changed, and there are literally dozens of aggregations and platforms from which to choose. But the question we continue to hear is “What’s the best way to get started?” Today’s VoIP Navigation Guide will help you make the right choices.

Before we begin, you need to do a little head-scratching yourself. Sit down with a pencil and paper (or a computer if you must) and jot down answers to our Top 10 Preliminary Questions:

  1. Is this for home or office use?
  2. How many simultaneous calls?
  3. How many users on the system?
  4. Will there be remote or traveling users?
  5. Is this a mission-critical system for you/others?
  6. What type & speed Internet service? Wi-Fi only?
  7. What is the skillset of those supporting the system?
  8. Do you want to babysit hardware for your system?
  9. What’s your initial and monthly budget for the project?
  10. What should happen to calls if your house/office burns down?

Skillset Matters! Let’s start with the obvious. The technical skillset of you and any other people that will be managing your VoIP server are critically important. This isn’t the old days where you only had to monitor people making long distance calls from within your own house. Once you connect a VoIP server to the Internet, anybody and everybody around the world can take a shot at your server and run up huge phone bills on your nickel unless you know what you’re doing or unless you deploy a server on which access is locked down to just you and trusted users and service providers.

We preach (regularly) that firewalls are essential if you’re going to deploy a VoIP server. In the home or office environment, that means that, in addition to your VoIP server, you also need a hardware-based firewall/router with no mapped ports to the VoIP server, period. Any other setup and it’s just a matter of time until you’re hacked.

In the hosted or cloud environment, it means at the very least a software-based firewall on your VoIP server with all access restricted to a whitelist of trusted users and providers. Any other setup and it’s just a matter of time until you’re hacked.

If you’re not qualified to manage either a hardware or software firewall, then your VoIP choices are limited. None of the major aggregations including PBX in a Flash, the FreePBX® Distro, AsteriskNOW, and Elastix provide any firewall protection as installed. While Fail2Ban is included, it is basically a log scanner which searches for failed login attempts and blocks IP addresses that make excessive login attempts. The major problem with Fail2Ban is that it takes time to run and, if your server is attacked from powerful servers, that may not happen until thousands of hack attempts have been executed.

We have attempted to address this problem with this summer’s new releases of Incredible PBX. In these new releases, whitelist access is locked down as part of the installation process. You have a choice of platforms.

On Cloud-based servers and depending upon your installation skills, we recommend:

On self-managed servers, you typically install the Linux operating system and then run the Incredible PBX installer. On smaller devices, we handle that for you. We recommend the following setups with the caveat that the old adage still applies: “You get what you pay for!” All four of the small hardware offerings below support WiFi-only operation. Just add the recommended WiFi USB dongle. For the CuBox-i, it’s built in. The VirtualBox setup takes less than 10 minutes.

Sizing Your Platform. Appropriate server and Internet capacity obviously turns on most of the answers you wrote down in the preliminary questionnaire. If the system will be used by less than a handful of people, you’re probably safe with the cloud-based solutions we’ve identified or one of the four low-cost devices listed above. Keep in mind that you need roughly 100Kbps of Internet bandwidth for each simultaneous VoIP call. If you have existing POTS lines from Ma Bell, those don’t consume Internet bandwidth but do consume local network resources. POTS line integration also requires additional hardware for each line. For less than 5 POTS lines, the OBi110 is an excellent choice. You’ll find it advertised in the right column of Nerd Vittles for under $50.

For up to a couple dozen low-call-volume employees, the RentPBX Cloud offering is a terrific bargain. It includes the necessary bandwidth not only to make calls but also to connect your extensions. When you get above those numbers of users or with heavy call volume, scaling matters. You don’t want to purchase a server only to discover on Day Two that it can’t handle the call volume. Here’s where the PBX in a Flash Forum can be a tremendous help. Describe your environment using the Top 10 Checklist from above. One of our hundreds of experts will lend a hand in recommending what you need to get started. Better yet, hire one of the gurus to handle the setup for you. It’ll save you thousands of dollars in headaches and easily pay for itself in future savings.

The PBX in a Flash Alternative. We haven’t mentioned PBX in a Flash as a solution for those just beginning their VoIP adventure. The reason is simple. The firewall is not preconfigured on PBX in a Flash, and somebody has got to do it unless your server is sitting behind a rock-solid, hardware-based firewall. The beauty of PBX in a Flash is that it’s incredibly flexible. You can choose not only the version of Asterisk and FreePBX to install, but you also can compile Asterisk with any collection of features desired. Once you get your feet wet with Incredible PBX, it’s our VoIP tool of choice, but it takes some skills on your part to run it safely. A good place to begin is the Nerd Vittles Quickstart Guide for PBX in a Flash 3. Enjoy!

Originally published: Wednesday, September 17, 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. Our forum is extremely friendly and is supported by literally hundreds of Asterisk gurus.



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

Another Ride on the Wild Side: Introducing Incredible PBX for CentOS 6.5 and 7

If you’re not one to turn down a new challenge and you consider yourself an open source purist, then this Bud’s for you. Last month we introduced the latest and greatest Ubuntu 14 offering with Incredible PBX. And today it’s RedHat’s turn with the brand new CentOS 7. Be forewarned that CentOS 7 is a very different Linux animal than CentOS 6.5, and it’s just two weeks old. RHEL 7 is only six weeks old. There are more than a few potholes in RedHat’s latest pathway to heaven. This results in a number of direct consequences in any Asterisk® and FreePBX® communications server which depends upon CentOS 7 under the covers. For openers, anything proprietary probably won’t work for a while. That includes Digium phones and Schmooze Com’s commercial modules for FreePBX. In addition, FreePBX 2.11 and 12 were designed using PHP 5.3. CentOS 6.5 is distributed with PHP 5.4. Ubuntu 14 and Fedora 20 have PHP 5.5. There are some incompatibilities between all three versions, and many of us still are sorting out what impact those incompatibilities will have on the overall reliability of the FreePBX platform and some of the Incredible PBX applications. You can help by testing this new build in a non-production environment. 95% of the feature set available in the CentOS 6.5 platform still works fine. But finding the gotcha’s is going to take some time… and some pioneers. So… roll up your sleeves and lend us a hand!

Incredible PBX™ for CentOS 6.5 and 7 is an independent aggregation that does not rely upon PBX in a Flash™ for its roots. Because of the nature of the CentOS platform, it was built from the ground up. PBX in a Flash will follow once the stability of the CentOS 7 platform has been demonstrated. The Incredible PBX installer is pure GPL2 open source code so you are more than welcome (encouraged!) to examine it, improve upon it, and share your discoveries with all of us.

Incredible PBX for CentOS 6.5 and 7 follows our new install procedure which means it’s up to you to first create a CentOS 6.5 or 7 platform. If you prefer Scientific Linux or Oracle Linux, feel free to start there. All work equally well as a base platform. Then you run the Incredible PBX installer. After 30-60 minutes of whirring, you’ll end up with an awesome (free) state-of-the-art Asterisk-based VoIP server with the very latest version of Asterisk 11 and FreePBX 2.11 as well as dozens of turnkey Incredible PBX applications. So enjoy a nice lunch while the Incredible PBX installer works its magic. No user intervention is required during the installation procedure. All text-to-speech (TTS) applications work out of the box. You can add Google’s Speech Recognition to many Incredible PBX applications by following our 5-minute tutorial.

Installing a Base CentOS Operating System

Let’s begin by installing 64-bit CentOS 7 or 6.5 on your favorite hardware or Desktop. Or you may prefer to use a Cloud provider1 that already offers a preconfigured CentOS 7 image. In the latter case, you can skip this section.

For those using a dedicated hardware platform or wishing to install CentOS 7 as a virtual machine, the drill is the same. Start by downloading the CentOS 7 minimal ISO or the 64-bit CentOS 6.5 minimal ISO. We recommend the Everything ISO at the moment since there currently is no minimal install ISO. Burn the whopping ISO to a DVD unless you’ll be booting from the ISO on a virtual machine platform such as VirtualBox. On virtual platforms, we recommend at least 1GB RAM and a 20GB dedicated drive. For VirtualBox, here are the settings:

Type: Linux
Version: RedHat 64-bit
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)
Start

Boot your server with the ISO, and start the CentOS 7 install. Here are the simplest installation steps:

Choose Language and Click Continue
Click: Install Destination (do not change anything!)
Click: Done
Click: Network & Hostname
Click: ON
Click: Done
Click: Begin Installation
Click: Root Password: password, password, Click Done twice
Wait for Minimal Software Install and Setup to finish
Click: Reboot

Configuring CentOS 6.5 or 7 for Incredible PBX Installation

Now log into your server as root and issue the following commands to put the basic pieces in place and to reconfigure your Ethernet port as eth0. Make a note of your IP address so you can log in with SSH.

setenforce 0
yum -y upgrade
yum -y install net-tools nano wget
ifconfig # figure out your server IP address here
sed -i 's|quiet|quiet net.ifnames=0 biosdevdame=0|' /etc/default/grub
grub2-mkconfig -o /boot/grub2/grub.cfg
reboot

If you’re on a virtual machine platform, now would be a good time to make an export or backup of your CentOS 7 image. The minimal install is about 500MB instead of 6.6GB. Don’t forget to first remove your hardware address (HWADDR) and network UUID from /etc/sysconfig/network-scripts/ifcfg-enp0s3 or whatever file name was assigned to your hardware. The saved image will be bootable with DHCP network support anywhere down the road.

NEWS FLASH: For those wanting to test things out using VirtualBox, a Scientific Linux 7 Minimal Install image (401MB) is now available on SourceForge. It gets you to right here in the install process.


Installing Incredible PBX for CentOS 6.5 or 7

Adding Incredible PBX to a running CentOS 6.5 or 7 server is a walk in the park. To restate the obvious, your server needs a reliable Internet connection to proceed. Using SSH (or Putty on a Windows machine), log into your new server as root at the IP address you deciphered in the ifconfig step at the end of the CentOS installation procedure above.

WARNING: If you’re using a 512MB droplet at Digital Ocean, be advised that their Ubuntu setup does NOT include a swap file. This may cause serious problems when you run out of RAM. Uncomment ./create-swapfile-DO line below to create a 1GB swap file which will be activated whenever you exceed 90% RAM usage on Digital Ocean.

Now let’s begin the Incredible PBX install. Log back in as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/incrediblepbx11.4.centos.tar.gz
tar zxvf incrediblepbx*
#./create-swapfile-DO
./IncrediblePBX*

Once you have agreed to the license agreement and terms of use, press Enter and go have a long cup of coffee. The Incredible PBX installer runs unattended so find something to do for the next 30-60 minutes unless you just like watching code compile. When you see “Have a nice day”, your installation is complete. Write down your admin password for FreePBX as well as your three “knock” ports for PortKnocker. If you forget them, you can reset your admin password by running /root/admin-pw-change. And you can retrieve your PortKnocker setup like this: cat /root/knock.FAQ.

Log out and back into your server as root and you should be greeted by something like this:

Or, if you started with a CentOS 6.5 or Scientific Linux 6.5 platform, you’ll see this:

1. Access the Asterisk CLI by typing: asterisk -rvvvvvvvvvv

2. Set Your Correct Time Zone by typing: /root/timezone-setup

3. Change ALL of Your Passwords by typing: /root/update-passwords

You can access the FreePBX GUI using your favorite web browser to configure your server. Just enter the IP address shown in the status display. The default username is admin and the password is what you wrote down or reset when the install completed. Now edit extension 701 so you can figure out (or change) the randomized passwords that were set up for your 701 extension and voicemail account: Applications -> Extensions -> 701. If you’re behind a hardware-based firewall, change the NAT setting to: YES.

Setting Up a Soft Phone to Use with Incredible PBX

Now you’re ready to set up a telephone so that you can play with Incredible PBX. We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of your server, 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 some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:


947 - Weather by ZIP Code
951 - Yahoo News
*61 - Time of Day
*68 - Wakeup Call
TODAY - Today in History

Now you’re ready to connect to the telephones in the rest of the world. If you live in the U.S., the easiest way (at least for now) is to set up a free Google Voice account. Google has threatened to shut this down but as this is written, it still works. The more desirable long-term solution is to choose several SIP providers and set up redundant trunks for your incoming and outbound calls. The PIAF Forum includes dozens of recommendations to get you started.

Configuring Google Voice

If you want to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX. If you want to use the inbound fax capabilities of Incredible Fax 11, then you’ll need an additional Google Voice line that can be routed to the FAX custom destination using FreePBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX 11. It’s free at least through 2013. Google Voice no longer is by invitation only so, if you’re in the U.S. or have a friend that is, head over to the Google Voice site and register.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it’s over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF
  • Call Options (Enable Recording)OFF
  • Global Spam FilteringON

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

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

Now you’re ready to set up your Google Voice trunk in FreePBX. After logging into FreePBX with your browser, click the Connectivity tab and choose Google Voice/Motif. To Add a new Google Voice account, just fill out the form. Do NOT check the third box or incoming calls will never ring!

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems.

Troubleshooting 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.

A Few Words about the Incredible PBX Security Model for CentOS 7

Incredible PBX for CentOS 7 joins last month’s Ubuntu 14 build as our most secure turnkey PBX implementation, ever. As configured, it is protected by both Fail2Ban and a hardened configuration of the IPtables Linux firewall. The latest release also includes Port Knocker for simple, secure access from any remote computer or smartphone. You can get up to speed on how the technology works by reading the Nerd Vittles tutorial. Your Port Knocker credentials are stored in /root/knock.FAQ together with activation instructions for your server and mobile devices. The NeoRouter VPN client also is included for rock-solid, secure connectivity to remote users. Read our previous tutorial for setup instructions. As configured, nobody can access your PBX without your credentials AND an IP address that is either on your private network or that matches the IP address of your server or the PC from which you installed Incredible PBX. You can whitelist additional IP addresses by running the command-line utility /root/add-ip. You can remove whitelisted IP addresses by running /root/del-acct. Incredible PBX is preconfigured to let you connect to many of the leading SIP hosting providers without additional firewall tweaking. We always recommend you also add an extra layer of protection by running your server behind a hardware-based firewall with no Internet port exposure, but that’s your call. And it’s your phone bill. :wink:

The IPtables firewall is a complex piece of software. If you need assistance with configuring it, visit the PIAF Forum for some friendly assistance.

Incredible Backup and Restore

We’re pleased to introduce our latest backup and restore utilities for Incredible PBX. Running /root/incrediblebackup will create a backup image of your server in /tmp. This backup image then can be copied to any other medium desired for storage. To restore it to another Incredible PBX 11 server, simply copy the image to a server running Asterisk 11 and FreePBX 2.11 and run /root/incrediblerestore. Doesn’t get much simpler than that.

Incredible PBX Automatic Update Utility

Every time you log into your server as root, Incredible PBX will ping the IncrediblePBX.com web site to determine whether one or more updates are available to bring your server up to current specs. We recommend you log in at least once a week just in case some new security vulnerability should come along.

In the meantime, we encourage you to sign up for an account on the PIAF Forum and join the discussion. In addition to providing first-class, free support, we think you’ll enjoy the camaraderie.

Incredible PBX: Pick Your Poison

We fully appreciate that Bleeding Edge technology isn’t right for everyone. Fortunately, with Incredible PBX, you have lots of options, and they’re all free. Come join the party and see what you’ve been missing.


Originally published: Monday, July 20, 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.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

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

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. :wink:

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.


 
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Some Recent Nerd Vittles Articles of Interest…

  1. 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. []

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