Category: Technology

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.


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.


 
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 $3.79 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. Did you happen to notice who is using someone else’s trademark in their brand name? But we digress. []

The Definitive VoIP Quick Start Guide: Introducing PBX in a Flash 3

It’s been an interesting year with RedHat’s acquisition of CentOS™. But the dust is slowly settling, and we’ve developed a new installation methodology for PBX in a Flash™ which we believe provides everyone with the best of all worlds. Like it or not, Red Hat® is in the driver’s seat now with CentOS, and Scientific Linux™ already has announced that they plan to fold into CentOS with the 7.0 release. That left the rest of us with two choices: fork CentOS and roll your own operating system or comply with the RedHat requirement to initially build a system with their ISO and then embellish it. The PBX in a Flash aggregation is just that. It’s always been built on a superset of the base CentOS operating system. That’s why we found the RedHat fanboy diatribes particularly offensive. PBX in a Flash has never provided a diluted or otherwise marginalized version of CentOS. If you don’t believe it, compare the list of RPMs on today’s build with the list on prior releases. They’re virtually identical even though (as you will see) the installation methodology is different. The bottom line is we don’t want to be in the operating system business, and the recent nightmare with OpenSSL should tell you why. Red Hat has a staff of hundreds to maintain RedHat and now CentOS. So why reinvent the wheel? When you peel away the marketing guys and the lawyers and the fan boys, that’s what open source has always been about. RedHat does what it does best, and we do the same. It never has meant you were getting a product that wasn’t genuine. You were getting a product that was embellished and enhanced to perform a specific task, telephony! By sticking with CentOS 6.5, we’ll all have a supported operating system on which to build telephony applications until the end of November, 2020. We can’t do better than that.

If you’re new to the VoIP community, we recommend you begin by watching this video. Before you begin the PBX in a Flash install procedure, you need to do three things first: pick your hardware platform, think about what types of phones you plan to use, and choose at least a couple of service providers to interconnect your PBX with the rest of the telephones in the world.


Making a Hardware Selection

We’re going to assume that you need a VoIP telephony solution that will support an office of up to several dozen employees and that you have an Internet connection that will support whatever your simultaneous call volume happens to be. This is above and beyond your normal Internet traffic. To keep it simple, you need 100Kbps of bandwidth in both directions for each call.1 And you need a router/firewall that can prioritize VoIP traffic so that all your employees playing Angry Birds won’t cause degradation in VoIP call quality. Almost any good home router can now provide this functionality. Remember to disable ALG on your router, and it’s smooth sailing.

For computer hardware, you’ll need a dedicated machine. There are many good choices. Unless you have a burning desire to preserve your ties with Ma Bell, we recommend limiting your Ma Bell lines to your main number. Most phone companies can provide a service called multi-channel forwarding that lets multiple inbound calls to your main number be routed to one or more VoIP DIDs much like companies do with 800-number calls.

If you’re building a system for home or SOHO use, you probably don’t need PBX in a Flash. If you want the same functionality for under $50 then go with a BeagleBone Black and add RasPBX and Incredible PBX. Our tutorial will show you how to do it. For the business model we’ve described above, any good dual-core Atom computer will suffice. You’ll find lots of suggestions in this thread. And the prices generally are in the $200-$400 range. For larger companies and to increase Asterisk’s capacity with beefier hardware, see these stress test results.

If your requirements involve retention of dozens of Ma Bell lines and complex routing of calls to multiple offices, then we would strongly recommend you spend a couple thousand dollars with a consultant. Some of the best in the business frequent the PBX in a Flash Forum, and they do this for a living. They can easily save you the cost of their services by guiding you through the hardware selection process. For business or for home, another alternative is available if you don’t want to babysit your own hardware. That’s a cloud-based solution such as RentPBX. For $15 a month, you don’t have to worry about electricity and a reliable Internet connection ever.

Choosing the Right Phones

If there is one thing that will kill any new VoIP deployment, it’s choosing the wrong phones. If you value your career, you’ll let that be an organization-driven decision after carefully reviewing at least 6-12 phones that won’t cause you daily heartburn. You and your budget team can figure out the price points that work in your organization keeping in mind that not everyone needs the same type of telephone. Depending upon your staffing, the issue becomes how many different phone sets are you and your colleagues capable of supporting and maintaining on a long term basis.

Schmooze Com has released their commercial End Point Manager (EPM) at a price point of $99 per server. They’ve been using the application internally to support their commercial customers for two years. If you’re doing a major installation, it’s the best money you will ever spend. Just sign up for an account with Schmooze to purchase the software. You can review the Admin User Guide here. The beauty of this software is it gives you the flexibility to support literally hundreds of different VoIP phones and devices almost effortlessly. Using a browser, you can configure and reconfigure almost any VoIP phone or device on the market in a matter of minutes. So the question becomes which phones should you show your business associates. That again should be a decision by you and your management and budget teams, but collect some information from end-users first. Choose a half dozen representative users in your company and get each of them to fill out a questionnaire documenting their 10 most frequent daily phone calls and listing each step of how they process those calls. That will give you a good idea about types and variety of phones you need to consider for different groups of users. Cheaper rarely is better. Keep in mind that phones can last a very long time, even lousy ones. So choose carefully.

The phone brands that we would seriously consider include Yealink, Digium, Snom, Aastra, Mitel, Polycom, Cisco, and Grandstream. Do you need BLF, call parking or multiple line buttons, a hold button, conferencing, speakerphone, HD voice, power over Ethernet support, distinctive ringtones for internal and various types of external calls, Bluetooth, WiFi, web, SMS, or email access, an extra network port for a computer, headset support, customizable buttons (how many?), quick dial keys, custom software, XML provisioning, VPN support? How easy is it to transfer a call? Do you need to mimic key telephones? Also consider color screens, touch screens, busy lamp indicators, extension modules (what capacity?). What do we personally use: Yealink’s T46G is our favorite, and we also have several Digium phones of various types, a couple of Aastra phones, a Grandstream GXP2200, a collection of Panasonic cordless DECT phones, a Samsung Galaxy S4 and Moto X connected through an OBi202 with an OBiBT Bluetooth Adapter, and a Samsung Galaxy S3 extension interconnected with Vitelity’s vMobile service to provide transparent connectivity on both WiFi and cellular networks. You can read all about vMobile here. It is the future of VoIP telephony.

Choosing VoIP Service Providers

One of the design differences between VoIP and the Ma Bell network that we’re all familiar with is that you no longer have to put all your eggs in one basket. The company or companies that you use to make outbound calls need not be the same as the ones you use to handle incoming calls. For home use, VoIP providers typically offer two types of plans: all-you-can-eat (which isn’t really) and pay-by-the-minute (which, in most cases, is priced by the fraction of the minute that you actually use the service). For business use, you have a choice of pay-by-the-trunk (each simultaneous call uses a trunk) and pay-by-the-minute (where you don’t have to manage your simultaneous calls). There was a third option over the past 5 years, and that was Google Voice which was free. But, good things don’t last forever, and Google is in the process of shutting down that service except for those that like making calls with a web browser. Hello, Ring.to.

For businesses, we strongly recommend that you stick with Ma Bell for your main business number only. That gets you listed in the phone book and provides 99.999% reliability for access to your business. Most phone companies can provide a service called multi-channel forwarding that lets multiple inbound calls to your main number be routed to one or more VoIP DIDs much like companies do with 800-number calls. For other business lines as well as home and SOHO setups, ditch Ma Bell as quick as you can. You’ll save boatloads of money. Give some thought to how much non-cellphone usage actually occurs in your situation. In many cases, you will find that pay-by-the-minute service for outbound calls is much less expensive than all-you-can-eat plans. Remember, there are no long term contracts on pay-by-the-minute services so try it and see what your usage habits actually are if you’re unsure. Keep in mind that acquiring inbound trunks for DIDs or phone numbers is almost always all-you-can-eat service ranging in price from $2-$8 a month. The PBX in a Flash Forum is chock full of recommendations. Just remember that, in doing your calculations, separate out the the time spent on incoming calls from the time spent placing outbound calls. Also keep in mind that redundancy is a luxury you never had in the Ma Bell days. Take advantage of it and sign up with multiple pay-by-the-minute providers for outbound (termination) service. You only pay for what you actually use. For inbound trunks, many providers offer failover service to different numbers if the primary connection dies. Even if the failover is to your cellphone, it beats missing the call. If international calling is a frequent part of your business or lifestyle, then spend some time exploring the options that are available. There are numerous all-you-can-eat solutions at incredibly affordable rates if you do your homework. Now let’s get started…

Installing CentOS 6.5

The new installation methodology for PBX in a Flash™ works like this. First, you’ll download the CentOS 6.5 server ISO for what is known as a minimal install. You still have your choice of 32-bit (339.7 MB) or 64-bit (417.3 MB) flavors. Burn the ISO to a USB Thumb Drive or a CD/DVD using a Mac or Windows machine.

If you’re building a system in the cloud or in a hosted environment, the base CentOS install usually has been done for you so you can skip this step.

If you’re using a dedicated PC or virtual machine with no operating system, boot from the CentOS 6.5 CD/DVD or ISO and go through the standard CentOS install procedure. Here are the CentOS 6.5 setup steps and entries that we recommend [in brackets] which will assure that your new server has wired network connectivity through DHCP and a non-LVM partition configuration which is easier to back up and restore. Don’t be intimidated by the list. The entire CentOS setup process only takes a minute or two.

1. Install or upgrade existing system
2. Test media [skip]
3. Begin setup [Next]
4. Choose language [English]
5. Keyboard [U.S. English]
6. Type Devices [Basic Storage Devices]
7. Discard Existing Data [yes]
8. Hostname [localhost.localdomain] ** BEFORE YOU CLICK NEXT, DO STEP 8a. **
  8a. Configure Network [Click eth0 & Edit. Check:Connect Automatically then Apply & Close]
9. Time Zone [New York] ** Uncheck: System Clock Uses UTC **
10. Root Password [** make it very secure **]
11. Type Installation: Create Custom Layout with Primary Partition checked for 11a and 11c
  11a. Create -> Standard Partition -> Mount Point: /boot Type: ext4 Size:200  Fixed
  11b. Create -> Standard Partition -> Mount Point: blank Type: swap Size:2048 Fixed
  11c. Create -> Standard Partition -> Mount Point: /     Type: ext4 Size:Fill to Max Size
12. NEXT
13. FORMAT
14. WRITE CHANGES
15. Checked: Install boot loader on /dev/sda  Boot loader CentOS List: /dev/sda3
16. Reboot when finished

Next, log in to your new server with your root credentials. First, check your disk partitioning to make sure everything looks okay: fdisk -l. Here’s what the partitioning looks like with a 20GB drive. For larger drives, your sda3 partition will obviously be larger.

Device    Boot Start   End  Blocks  ID System
--------- ---- ----- ----- -------- -- ----------
/dev/sda1   *      1    26   204800 83 Linux
/dev/sda2         26   287  2097152 82 Linux swap
/dev/sda3        287  2650 18979840 83 Linux

Now let’s prepare your server for installation of PBX in a Flash 3. None of these three commands will do any damage if your server happens to already be configured properly.


sed -i 's|no|yes|' /etc/sysconfig/network-scripts/ifcfg-eth0
ifup eth0
yum -y install wget nano

Installing PBX in a Flash

Now let us welcome you to the World of PBX in a Flash™. This is our best release ever whether you’re a total newbie or an experienced Asterisk developer. You can’t really appreciate what goes into an open source product like PBX in a Flash until you try doing it yourself. If you want to actually learn about Asterisk from the ground up using pure source code to customize your VoIP deployment, PBX in a Flash has no competition because your only other option is to roll your own starting with a Linux DVD. So our extra special kudos go to Tom King, who once again has produced a real masterpiece in that it is very simple for a first-time user to deploy and, at the same time, incredibly flexible for the most experienced Asterisk developer. The new PIAF3™ release not only provides a choice of Asterisk and FreePBX versions to get you started. But now you can build and deploy standalone servers for SugarCRM™, NeoRouter™ VPN, YATE™, FreeSwitch™, and OpenFire™ XMPP using the standard PIAF3 installer. So let’s get started.

Now we’re ready to begin the PIAF3 install. Issue the following commands to get started:

cd /root
wget --no-check-certificate http://nerd.bz/pbxinaflash3
mv pbxinaflash3 piaf3-install.tar.gz
# alternate site below if SourceForge is down
# wget http://pbxinaflash.com/piaf3-install.tar.gz
tar zxvf piaf3-install.tar.gz
./piaf3-install

When the install begins, there’s a 5-10 minute process to reconfigure CentOS by adding over 500 applications to the base install. Be patient. When it completes, your server will reboot, and you’re ready to begin the PBX in a Flash installation process. Choose option A to continue with the installation. While PBX in a Flash supports a number of versions of Asterisk and FreePBX, we believe the combination of Asterisk 11 and FreePBX 2.11 is so compelling in terms of functionality, stability, and security that the other options are no longer worth considering. We wholeheartedly recommend choosing PIAF-Green with FreePBX 2.11 as your platform.

For today, we’re installing PBX in a Flash. So leave it highlighted, tab to OK, and press Enter.

Now pick your PIAF flavor, tab to OK, and press Enter. HINT: Green is the fourth option. :-)

The PIAF Configuration Wizard will load. Press Enter to begin.

Unlike any other aggregation, PIAF gives you the opportunity to fully configure Asterisk using make menuconfig if you know what you’re doing. For everyone else, type N and then confirm your choice.

Next, you’ll need to choose your Time Zone again for PHP and FreePBX. Don’t worry if yours is missing. A new timezone-setup utility is also available to reconfigure this to any worldwide time zone once the install has completed.

Next, choose your version of FreePBX to install. As we said, we recommend FreePBX 2.11. Note that Incredible PBX 11 requires PIAF-Green and FreePBX 2.11.

Finally, you need to choose a very secure maint password for access to FreePBX using a browser. You can pick your own, or the installer will generate one for you. Don’t forget it.

The installer will give you one last chance to make changes. If everything looks correct, press the Enter key and go have lunch. Be sure you have a working Internet connection to your server before you leave. :wink:

In about 30-60 minutes, your server will reboot. You should be able to log in as root again using your root password. Write down the IP address of your server from the status display (above) and verify that everything installed properly. Note that Samba is disabled by default. If you want to use your server with Windows Networking, run configure-samba once your server is up and running and you’ve logged in.

If you’re familiar with Asterisk and FreePBX, then you can take it from here. You now have a fully functioning platform on which to create your latest VoIP masterpiece. If you’re new to all of this, keep reading…

Configuring PBX in a Flash

Most PIAF Configuration is accomplished using the FreePBX Web GUI. Point your browser to the IP address shown in the status display above to display your PIAF Home Page. Click on the Users tab. Click FreePBX Administration. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in the Config Module phase of the PBX in a Flash installation procedure above.

Here’s a quick overview of what needs to happen before you can start making and receiving calls. You’ll need an account with at least one phone number for people to call you (known as a DID), and you’ll need an account to place outbound calls to plain old telephones throughout the world. Our Vitelity DID deal at the bottom of this article is a terrific service, and Vitelity also provides tremendous financial support to both the Nerd Vittles and PBX in a Flash projects. For outbound calling, you also can use Vitelity or choose from the provider recommendations on the PIAF Forum.

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 PBX in a Flash. Extensions talk to trunks 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 the phones actually ring. In a nutshell, that’s how a PBX works. There are lots of bells and whistles that you can explore down the road. FreePBX now has some of the best documentation in the business. Start here.

To get a minimal system functioning to make and receive calls, here’s the 2-minute drill. Create at least one extension with voicemail. Next, configure a trunk to handle your outside calls. Then set up inbound and outbound routes to manage incoming and outgoing calls. Finally, add a telephone or softphone with your extension credentials.

If this sounds like Greek to you, then install Incredible PBX 11. It’s a 5-minute task. Incredible PBX does all the heavy lifting for you by configuring an extension, building dozens of trunks for the major SIP providers, and creating default routes to manage your calls. You also get a terrific collection of utility programs for Asterisk that handle everything from telephone reminders and wakeup calls to weather and news reports. To get started, log into your server as root and issue the following commands. Then jump to the Incredible PBX 11 tutorial and continue your journey there.

cd /root
wget http://incrediblepbx.com/incrediblepbx11.gz
gunzip incrediblepbx11.gz
chmod +x incrediblepbx11
./incrediblepbx11

A Few Words About Security. PBX in a Flash has been engineered to run on a server sitting safely behind a hardware-based firewall with NO port exposure from the Internet. Leave it that way! It’s your wallet and phone bill that are at stake. If you’re running PBX in a Flash in a hosted environment with no hardware-based firewall, then immediately read and heed our setup instructions for Securing Your VoIP in the Cloud Server. DO NOT RUN PBX IN A FLASH IN THE CLOUD WITHOUT INSTALLING AND ACTIVATING THE IPTABLES FIREWALL. HINT: TRAVELIN’ MAN 3 WILL DO THE HEAVY LIFTING FOR YOU. We would encourage you to visit your PIAF Home Page regularly. It’s our primary way of alerting you to security issues which arise. You’ll see them posted (with links) in the RSS Feed shown above. If you prefer, you can subscribe to the PIAF RSS Feed or follow us on Twitter. For late-breaking enhancements, regularly visit the Bug Reporting & Fixes Topic on the PIAF Forum. Enjoy!

Originally published: Wednesday, May 28, 2014




Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID 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 $3.79 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…

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

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.


 
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…

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




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.


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.


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.


 
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. 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


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.


 
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…

4 Months in Paradise: The Return of Free International VoIP Calling

With the impending implosion of Google Voice, it seemed appropriate to begin our quest for alternative termination providers. One of the real beauties of VoIP technology is you don’t have to put all of your eggs in one basket particularly in the termination department. It costs almost nothing to set up accounts with multiple providers for outbound calling. In addition to redundancy, the other clear advantage in using multiple providers for outbound calls is that you can take advantage of special rates to different destinations. So here’s the bargain of the week. If you have loved ones traveling to South America, Europe or Asia this summer, now’s your chance to sign up for VoIP service with FreeVoipDeal and enjoy four months of free calling to more than 50 countries around the world for every $15 of credits you purchase on their web site. Please note the fine print: “FreeVoipDeal reserves the right after a certain amount of calls to start charging the default rate.” There is no mention of what that “certain amount” happens to be. When your free calling finally ends, you can either purchase $15 of additional credits for 120 more “free” days or continue to call all of the previously free destinations for about 2¢ a minute.

The company behind FreeVoIPDeal is betamax which hosts over 30 sites offering varying deals to different countries. BEWARE: The prices change regularly. So a country that’s free today may suddenly cost money tomorrow. How does a mere mortal keep track? Well, betamax probably hopes that you won’t. But an enterprising individual named Robert Siemer has done the work for you. His backsla.sh/betamax web site automatically updates the pricing for all betamax sites every day! If this sounds like a lot of work to save a few cents a minute, you’d be right. And Vitelity which sponsors both the Nerd Vittles and PBX in a Flash projects offers consistently low rates to all of these countries. You’ll find a DID special at the end of this article, and their excellent international rate table is available at this link.

Setting Up an Account. Before you can set up a trunk in PBX in a Flash, you’ll first need to create a FreeVoipDeal account. In the “old days” this required use of their Windows client to obtain your credentials. Now you can simply create an account on the web site at this link. You’ll need either a regular land line or a cell phone number to verify your registration. Once you’re set up and you’ve deposited at least 10 euros (about $15) in your account, it’s time to set up a SIP trunk and outbound route in PBX in a Flash.

Configuring a Trunk with PBX in a Flash. Assuming you already have a phone registered to an extension in PBX in a Flash, it’s a one-minute drill to configure a trunk and outbound route to support FreeVoipDeal. Using a browser, log into FreePBX® using your maint username and password. Choose Connectivity -> Trunks -> Add SIP Trunk. Name the trunk: FreeVoipDeal. For the Dialed Number Manipulation Rules, enter Prepend: 1 and Match Pattern: NXXNXXXXXX. Clear out all of the default entries in Outgoing and Incoming Settings. Then, in Outgoing Settings, enter Trunk Name: freevoipdeal. For the PEER Details, enter the following using your actual account USERNAME and PASSWORD. Then SAVE your settings and reload FreePBX.

username=USERNAME
authuser=USERNAME
secret=PASSWORD
type=peer
qualify=yes
nat=yes
insecure=port,invite
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
dtmfmode=auto
disallow=all
canreinvite=no
allow=ulaw

There’s no need to enter a CallerID number. All of the outgoing calls will be delivered as ANONYMOUS. You also won’t need to register with the provider since Asterisk® can handle this on the fly using your credentials entered above.

Configuring an Outbound Route with PBX in a Flash. One more step, and you’ll be ready to start making calls. Choose Connectivity -> Outbound Routes. For the Route Name, enter: FreeVoIPDeal. For the Dial Pattern to make U.S. calls, enter: NXXNXXXXXX. If you want to force callers to dial a prefix to use the FreeVoipDeal trunk, then enter a 9 or some other number in the Prefix field. For Trunk Sequence 0, choose: FreeVoipDeal. Click Submit Changes and restart FreePBX when prompted. You’re done!

Making Your First Call. Using a phone or softphone logged into your server, dial the prefix (if any) plus the 10-digit number of someone in the United States. When the called party answers, make sure you can hear the called party and vice versa. If not, open Settings -> SIP Settings in FreePBX and add your External IP and Local Network settings. Also make certain the NAT entry is set to YES.

Configuring Your Server for International Calls. We do not recommend configuring your server to permit international calls to everywhere. The reason is simple. If strangers manage to access one of your extensions, they can run up your phone bill in a hurry. For this reason, we also strongly recommend that you do not configure automatic credit card replenishment with any VoIP provider!

For international calling, we recommend you add a separate Dial Pattern to both your FreeVoipDeal trunk AND the outbound route for each country code you wish to enable. Here is the complete list of codes. For example, to allow calls to Germany from another country, you’d add 49XXXXXXXXXX, save your changes, and reload FreePBX.

Spoofing Your CallerID. If you first verify that you own a number by using the web portal, you then can spoof the outbound CallerID using the number you verified. Just add the following entries to your trunk settings replacing 9991234567 with your verified CallerID number. Special thanks to @hillclimber on the PIAF Forum for the tip.
fromuser=0019991234567
sendrpid=yes

Originally published: Friday, April 25, 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…

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