Tag Archives: piaf

#83329085 / gettyimages.com

2015 is starting off with lots of surprises for the VoIP community so let’s get right to it. Sangoma Technologies has purchased Schmooze Com with all its assets including FreePBX® on January 1. You can read all about it here and here. Please do. The bottom line is the ownership of FreePBX has changed, but the development staff and presumably the future direction of the project have not. As usual, there is more than a little bad mouthing of Fonality for the direction it took the trixbox project while promising to be "different" with this acquisition. We hope so. Keep reading for the rest of the story…

We’ve known the original developers of FreePBX since the Asterisk Management Portal days. And the same goes for the Asterisk@Home and trixbox project team as well as the current FreePBX development team. When we began the PBX in a Flash project, the very first financial backer of our project was Sangoma, and their support of the open source community has been unwavering. What follows is a wakeup call that all is not well in the FreePBX community, and now Sangoma is in a position to fix it. We hope they will… and soon!

When Schmooze Com decided to discontinue its commercial PBX offering and roll it into commercial modules for FreePBX, we were one of the early testers and supporters of those modules and the new approach. We also had an ongoing discussion with Tony Lewis regarding patents, copyrights, commingling of commercial modules with open source code, and numerous other topics. The objective for us and for Tony was to develop a long-term strategy for Schmooze Com that would assure commercial viability while protecting the open source character of FreePBX. In exchange for including commercial module support in the PBX in a Flash offerings, Schmooze Com agreed to build a web site that could detect the platform of the user so that a portion of the proceeds of the commercial purchases could be returned to our project to fund our development efforts. We never saw a dime!

During this same period, we also were seeking a commercial VoIP provider to provide commercial-quality technical support for PBX in a Flash users whenever the need arose. Schmooze Com seemed like a natural fit given our joint development efforts. In May of 2012, we entered into a partnership arrangement with Schmooze Com, a copy of which is reproduced below:

Support and commercial module development continued uneventfully through the end of 2012 with checks to the PBX in a Flash project tallying up to less than $1,000. That just meant our users didn’t have many problems, or so we thought. On January 10, 2013, we received the following email from Tony… but no check:

We have been tracking down some weird issues with a few modules in PBXiaF and have it tracked down that your sysadmin RPM is really old.

Because that RPM is always changing we have created a new REPO that only contains the 3 needed RPMS for commercial module support.

Can you include this repo in your upgrade scripts and next build instead of relying on updating your repos when we change the RPMS

We will always keep this repo updated with the RPMS needed for commercial modules

A week later, we received a follow up email… but no check:

We now have our Portal setup to track Commercial Modules on a per system type basis so we can start paying you a commission on PBXiaF systems.

We seem to keep having issues with PBXiaF users not having updated RPMs such as sysadmin.

We have setup a repo that we would like you to include that way they are pulling the needed RPMs from our repo. Its [sic] the same repo we are now using in FreePBX and Asterisk Now is now also using.

We made the necessary changes to PBX in a Flash and incorporated the Schmooze commercial repo based upon the assurance that it would only "contain the 3 needed RPMS for commercial module support." This is critically important from a security standpoint since any repo activated on a Linux server basically gets a blank check with root privileges to modify virtually anything on that server. Keep reading! It gets worse.

In February, 2013, Schmooze Com acquired FreePBX from Bandwidth.com. Perhaps not coincidentally, that also marked the end of the money trail from Schmooze Com to the PBX in a Flash project. Shortly thereafter, we began receiving reports from various PIAF users that their (paid) call for commercial technical support was more of a sales pitch urging them to switch to the FreePBX Distro for "better support." Compare that advice to Section 5 of the Memorandum of Understanding we have reproduced above.

In 2014, our relationship with Schmooze Com went from bad to worse as the company began squeezing other contributors to the PBX in a Flash project for money. One provider of SIP services developed an add-on open source module which end-users could download and install into FreePBX to facilitate configuration of their SIP credentials. This provider, who also happened to be a competitor of Schmooze Com’s SipStation, received a threatening email in March of 2014 which included the following:

We also see you have a FreePBX module that is used to manage and configure your trunks which violates our Copyright Policy on using the FreePBX Framework and module system. As stated on our trademark page.

"FreePBX provides a module system to allow plugging in 3rd party modules into your FreePBX system. Any module that uses the FreePBX Module, Framework or GUI system must be released as GPL and use of the module must be for controlling or managing other GPL or open source software. Schmooze Com, Inc as the copyright holder does reserve the right to release modules that are not GPL and under a different license under a dual license model."

Since you [sic] modules sole purpose is to configure and manage your trunking service this would be in violation of FreePBX usage policy.

Imagine the reaction from Sangoma if Digium had ever announced that Asterisk modules to support analog cards from suppliers other than Digium could not be used with Asterisk because it would violate Digium’s "Copyright Policy on using the [Asterisk] Framework and module system."

Shortly thereafter, a number of cloud service providers contacted us indicating that Schmooze Com was demanding royalties for use of the open source FreePBX product in cloud offerings of the open source PBX in a Flash product line. Never mind that Schmooze Com uses hundreds of open source products commercially including Asterisk, Apache, PHP, and MySQL without payment of any license fees.

Get the picture? Now mere use of the open source FreePBX product in a commercial offering was prohibited without payment of a Schmooze Com "trademark and copyright fee." Now tell me again that yarn about Fonality being a lousy steward of the trixbox project. They never pulled a stunt like this! And then, of course, there’s the plain language of the FreePBX GPL license:

1. You may copy and distribute verbatim copies of the Program’s source code as you receive it, in any medium, provided that you conspicuously and appropriately publish on each copy an appropriate copyright notice and disclaimer of warranty; keep intact all the notices that refer to this License and to the absence of any warranty; and give any other recipients of the Program a copy of this License along with the Program.

You may charge a fee for the physical act of transferring a copy, and you may at your option offer warranty protection in exchange for a fee.

2. You may modify your copy or copies of the Program or any portion of it, thus forming a work based on the Program, and copy and distribute such modifications or work under the terms of Section 1 above, provided that you also meet all of these conditions:

a) You must cause the modified files to carry prominent notices stating that you changed the files and the date of any change.
b) You must cause any work that you distribute or publish, that in whole or in part contains or is derived from the Program or any part thereof, to be licensed as a whole at no charge to all third parties under the terms of this License.
c) If the modified program normally reads commands interactively when run, you must cause it, when started running for such interactive use in the most ordinary way, to print or display an announcement including an appropriate copyright notice and a notice that there is no warranty (or else, saying that you provide a warranty) and that users may redistribute the program under these conditions, and telling the user how to view a copy of this License. (Exception: if the Program itself is interactive but does not normally print such an announcement, your work based on the Program is not required to print an announcement.)

The final straw (as if we needed one) was the recent declaration that FreePBX commercial modules "are not Open Source GPL and are only designed to work with the FreePBX Distro." This, of course, is long after many PBX in a Flash users had purchased commercial modules on the frequent recommendation of Schmooze Com employee postings on the PIAF Forum.

And to start the new year off with a bang, Schmooze Com quietly added additional (non-commercial) components to their commercial repository which immediately broke the Fail2Ban security module used by PBX in a Flash. Through the commercial module repo, we now have a backdoor security issue because Schmooze Com is no longer honoring their agreement to restrict the Schmooze Com commercial repo to "the 3 needed RPMS for commercial module support."

We will fix it shortly… and permanently.

Ultimately you, our readers, get to judge whether Schmooze Com’s stewardship of the FreePBX project has been a model for the open source community. From our vantage point, it has been anything but that. Sangoma has enormous good will in the open source community. We trust they will take the necessary steps to correct these abuses for the benefit of the open source FreePBX project and those who continue to develop and use it.

Continue reading Page 2…

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Originally published: Monday, January 12, 2015

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

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Some Recent Nerd Vittles Articles of Interest…

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.

• SIP Gateways for Google Voice — GVsip and Simonics

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


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.

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

The recommended platform is CentOS or Scientific Linux. Start here:

sed -i 's|no|yes|' /etc/sysconfig/network-scripts/ifcfg-eth0
ifup eth0
setenforce 0
yum -y upgrade
yum -y install net-tools nano wget
ifconfig # to 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
# for CentOS/Scientific Linux 6.5/6.6 only, perform these additional steps:
wget http://pbxinaflash.com/update-kernel-devel
chmod +x update-kernel-devel
./update-kernel-devel
reboot

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

cd /root
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. For the time being, type Y. When the menuconfig menu displays during the install, type X to save your settings and exit. No changes are required.

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.

In about 30-60 minutes, your server will reboot. You should be able to log in as root again using your root password.

Because of a version update to PEAR that is not supported by FreePBX, you’ll need to issue the following commands to clean things up: [NOTE: This has been resolved in latest PIAF3 releases.]

chattr -i /usr/bin/pear
chmod +x /usr/bin/pear
amportal restart
status

We also strongly recommend that you immediately upgrade your version of Asterisk to the current release. If you’re using PIAF-Green with Asterisk 11, we have a script that will do the heavy lifting for you: [NOTE: This already has been addressed in latest PIAF3 release.]

cd /root
wget http://pbxinaflash.com/upgrade-asterisk11-piaf.tar.gz
tar zxvf upgrade-asterisk11-piaf.tar.gz
rm upgrade-asterisk11-piaf.tar.gz
./upgrade-asterisk-piaf

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!

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Originally published: Wednesday, May 28, 2014 Updated: Wednesday, December 3, 2014

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

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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.

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Special Thanks to Our Generous Sponsors

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

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

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

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

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

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