Posts tagged: freepbx

Freedom and the FreePBX Cloud: Is an Apple-like Ecosystem GPL-Compliant?

Short Answer: No way, José!     Right Answer: Sangoma should fix it.     Our Answer: New GPL Repo fixes it… today!

We began our series on FreePBX® by providing a GPL-compliant alternative to the base design of the FreePBX GUI minus the elements which have made redistribution and/or code modification difficult despite the clear language of the product’s GPL licenses. In our last article, we introduced new turnkey versions of Incredible PBX for CentOS featuring your choice of the 2.11 or 12.0 Incredible PBX GUI. Coming soon will be new releases of Incredible PBX for the Ubuntu, Debian, and Raspbian platforms so hang in there.

This week we begin our examination of the actual FreePBX design and the morphing that has taken place. We want to give you the full picture of why this led to our decision to no longer support the FreePBX approach to “GPL” software design. We also will provide some additional GPL tools that open up the platform in the way the GPL license requires.

It’s important for everyone to understand the impact of commercialization on project development when organizations bend the rules to suit their own commercial purposes. None of this was Sangoma’s doing. But FreePBX is now Sangoma’s GPL project, and it’s up to them to clean up the mess. For openers, nobody forced the FreePBX developers to release the FreePBX code with a GPL license. But they did it… almost 10 years ago! Only after the product became hugely popular did these folks apparently conclude that maybe giving away their software wasn’t such a good idea after all. You can track when the wheels came off the bus by looking at the project’s history on SourceForge. Not surprisingly, it coincides with SchmoozeCom’s entry into the picture. As Richard Stallman of the Free Software Foundation would tell you, this isn’t about whether code is open source software. Some FreePBX modules are and many are not. But providing source code is merely one aspect of the GPL. So let’s start with some of the actual language from the GPL license:

When we speak of free software, we are referring to freedom, not price. Our General Public Licenses are designed to make sure that you have the freedom to distribute copies of free software (and charge for them if you wish), that you receive source code or can get it if you want it, that you can change the software or use pieces of it in new free programs, and that you know you can do these things.

To protect your rights, we need to prevent others from denying you these rights or asking you to surrender the rights. [Emphasis added.]

Today we want to cover the first of several topics you won’t ever hear about in a (commercial) “advanced” training class for FreePBX. In case you haven’t attended one of these lovefests, the training is intended to let (paying) students learn how to customize the settings of the GUI for others willing to pay someone to build them a PBX. There’s nothing particularly wrong with that unless you believe everything associated with free software should be free. We don’t. In any case, you’ll learn how to create extensions and ring groups, inbound and outbound routes, trunk setups, and many of the other (basic) things that Nerd Vittles has been covering (for free) for years. And, of course, you will learn how to market the FreePBX brand and Sangoma-produced commercial modules.

What you won’t hear is anything about the inner workings of FreePBX much less how to customize the product for your own use, i.e. the types of modifications envisioned by the clear terms of the GPL. Those GPL “features” are available on a per customer basis for substantial “customization fees.” Translation: roughly the same cost as a new Hyundai for your kid headed off to college. And there’s one other hidden surprise. Even with custom branding of FreePBX, you will remain a captive in the so-called FreePBX ecosystem.

If you’ve enjoyed Apple’s App Store approach to system lock-in, then you’ll feel right at home with FreePBX. The wrinkle is that the FreePBX approach is even more restrictive than Apple’s. For openers, anyone wishing to sell their own commercial module need not apply. Unlike Apple, no commercial offerings from anyone but Sangoma are permitted in the FreePBX ecosystem. Imagine if Digium had adopted a similar approach by barring modules from competing hardware companies from interfacing with Asterisk®. Where would that have left Sangoma? In the case of FreePBX, even if you want to give away a FreePBX-compatible GPL module, you’re out of luck with FreePBX 12 unless you’re willing to underwrite Sangoma’s unlimited legal expenses if they ever get sued. Note our emphasis on unlimited. Sangoma claims they merely copied a general indemnification provision used by others such as Rackspace. But, as one of our readers pointed out:

The link that they claim they used as a template is one I would sign. Sangoma reworded things so that ALL liability is yours, even if an issue arises in their code that affects your code (after the fact). Sangoma in that case, is responsible but YOU have to pay for their legal fees. You cannot have a final say in settlements, they do. They can select whatever priced attorneys they want (you have no say). There is no ‘reasonable’ word usage. They dropped it.

As for your GPL module, yes, you can manually load it and run it without signing the indemnification agreement, but users will have to endure nasty warnings and emails every day which suggest that their server has been compromised.1 Apple, on the other hand, screens free and non-free additions to their App Store and includes literally thousands of third-party apps without anyone having to pay Apple’s legal fees. FreePBX proclaims that “Free Stands for Freedom” but…

I’m reminded of a book that was published during the Vietnam War era: “Military Justice is to Justice As Military Music is to Music.” If this is Sangoma’s idea of freedom, I’m not quite sure why anyone would want it except for the fact that they’re the only GUI game in town. The Sherman Act may be unfamiliar territory in Canada, but it might be worth a careful look.

Here’s where the GPL breaks down. Despite the best of intentions, the GPL drafters believed that handing someone the source code for a program was the best way to insure freedom to redesign and redistribute computer programs. That works well when the computer program is a couple hundred lines of code, but it breaks down quickly when you’re dealing with a program that’s been commingled with a commercial Cloud-based hosting service shrouded in secrecy and you’re staring at a million lines of code that can best be described as “engineered obfuscation.” Think of it as handing someone a plate of your grandma’s cookies and, when asked for the recipe, you say, “All of the ingredients are right there in front of you.” Yes, but…

This is a critically important point so let’s cover it in the context of FreePBX. What do you get and what do you not get when you install or use the product? Because the FreePBX GPL modules are written in unencrypted PHP code, you automatically get the source code when you install each module. It used to be that you also could acquire the modules on a public web site provided by the developers, now Sangoma. As noted last week, that openness came to a screeching halt with FreePBX 12. Until our repository was made available, you could scour the web high and low, but you wouldn’t find the GPL “free” modules for FreePBX 12 in a format directly usable by the FreePBX GUI and its Module Admin update feature which is perhaps the best feature of the GUI. In fact, until today, the only way to acquire the modules in a usable format with error correction was through the FreePBX GUI interface itself using the proprietary, hidden “ecosystem” maintained by Sangoma. The acquisition process itself is buried deep in a million lines of spaghetti code. Yes, you can get the source code, but…



Sangoma hopefully will ponder the words of Richard Stallman, the Founder and President of the Free Software Foundation:

Clearly that server does not respect our freedom, and we should refuse to use it, for the most part.

If we use a GUI for PBX’s, we should load our modules in some way that treats us decently.

So why the mystery with acquisition of FreePBX modules? The simple is answer is that it restricts everyone’s freedom. You can’t redistribute FreePBX without keeping Sangoma and the “non-free” FreePBX ecosystem in the middle of the equation. This provides the ongoing platform for Sangoma to peddle the sale of (only) their branded SIP trunking service as well as (only) their commercial modules. This may be their idea of freedom, but…

Last week we provided the first glimpse of freedom providing a means to break away from the trademark gimmicks of the mothership by using our reengineered GPL GUI with our repository of GPL modules for the new product. What you still lacked was the freedom to break away from our universe and go your own way. Why? Because the FreePBX developers have never revealed their Cloud’s secret sauce much less the tools necessary to create your own GPL module repository and have it function properly within the GUI. Without the cloud access and control, you lose the key module update and monitoring capabilities of the product itself plus the ability to upgrade the GUI to a later version. We used to call this CrippleWare, software with only limited functionality unless you cough up the big bucks. They’ll tell you that it’s all in the source code…

Well, not quite all. FreePBX is open source GPL software minus the secret sauce hidden in Sangoma’s Cloud which is the antithesis of the freedom component of the GPL. If you don’t appreciate the difference and why this runs counter to the GPL, read Richard Stallman’s explanation here. Because Cloud access by design is the only means provided in the FreePBX GUI to load new GPL modules, or to check for and update existing modules, or to upgrade the FreePBX GUI itself,2 the Cloud component is clearly an integral component of FreePBX. As such, it also must be licensed under the GPL and all its source code made available. In the words of the Free Software Foundation:

I’d like to incorporate GPL-covered software in my proprietary system. Can I do this?

You cannot incorporate GPL-covered software in a proprietary system. The goal of the GPL is to grant everyone the freedom to copy, redistribute, understand, and modify a program. If you could incorporate GPL-covered software into a non-free system, it would have the effect of making the GPL-covered software non-free too.

A system incorporating a GPL-covered program is an extended version of that program. The GPL says that any extended version of the program must be released under the GPL if it is released at all. This is for two reasons: to make sure that users who get the software get the freedom they should have, and to encourage people to give back improvements that they make.

However, in many cases you can distribute the GPL-covered software alongside your proprietary system. To do this validly, you must make sure that the free and non-free programs communicate at arms length, that they are not combined in a way that would make them effectively a single program.

The difference between this and “incorporating” the GPL-covered software is partly a matter of substance and partly form. The substantive part is this: if the two programs are combined so that they become effectively two parts of one program, then you can’t treat them as two separate programs. So the GPL has to cover the whole thing.

If the two programs remain well separated, like the compiler and the kernel, or like an editor and a shell, then you can treat them as two separate programs—but you have to do it properly. The issue is simply one of form: how you describe what you are doing. Why do we care about this? Because we want to make sure the users clearly understand the free status of the GPL-covered software in the collection.

If people were to distribute GPL-covered software calling it “part of” a system that users know is partly proprietary, users might be uncertain of their rights regarding the GPL-covered software. But if they know that what they have received is a free program plus another program, side by side, their rights will be clear.

Of course, every new module release brings a new opportunity to change the file and directory structure hidden in the Cloud to once again disguise the secret components required for proper GUI operation. Trust us. They have. Why else would you change a file name from modules-12 to all-12 except to conceal its identity? It’s called security through obscurity. Try searching your server for all-12 and see what you find. This hidden file is what locks you into the Sangoma commercial ecosystem. They call it freedom. It’s really anything but that. A more descriptive label would be a hidden, proprietary GOTCHA. You get some of the source code to make FreePBX work properly, but…

Building an Independent GPL Cloud Repository for the Incredible PBX GUI

Today we’re going to fix this deficiency at least for those using the new Incredible PBX GUI by offering independently developed GPL code that provides the freedom to build your own Cloud-based ecosystem should you wish to do so. We would encourage Sangoma to do the right thing. Stop listening to the former owners of the FreePBX project and become a good GPL steward. It’s your project now. You’ve owned it for almost six months! You’re also a better company than the one you bought. So start acting like it. Bring the FreePBX Cloud-based components out into the open and provide the tools necessary to use them as your GPL product license requires.

What we are providing today are all the components necessary to build an independent GPL Cloud that is compatible with the Incredible PBX GUI. This includes a base install of existing GPL modules that are compatible with versions 2.11 and 12 of FreePBX plus the toolkit to maintain an independent GPL Cloud. To load future modules and updates into your repository, you’ll need a Linux LAMP server running the latest version of Apache and at least PHP 5.4. Neither Asterisk® nor FreePBX is required on the server platform. Be advised that CentOS 6.5 and 6.6 ship with PHP 5.3 so you’ll need to perform the following steps to bring your server up to the 5.4 or 5.5 version of PHP before proceeding. Be advised that your GPL Cloud will only work with GPL-licensed versions of Incredible PBX running the 2.11 or 12 release of Incredible PBX GUI. See last week’s tutorial to get started.

Before we begin, several cautionary notes are in order. First, we can’t control Sangoma’s behavior. Assuming they decide not to comply with the GPL by keeping their Cloud service proprietary, a simple tweak on their end could change the location of their Cloud’s secret sauce at any time. That could very well break the ability to download future GPL modules from their repositories using this toolkit. But don’t worry. If that happens, we’ll be the first to let you know. We figured it out once, and we can figure it out again. You can run, but you cannot hide! We’ll also show you an alternative method to load new modules into your own repository. Second, don’t even think about using your own repository while retaining the original FreePBX GUI instead of updating to the Incredible PBX GUI. A single module update on their end could do a couple of things. It could overwrite the location of the module repositories and restore theirs. Or it could completely disable your server after detecting that you had changed the internal workings of FreePBX. Remember when Apple did just that with jailbroken iPhones? We’re not suggesting Sangoma would actually pull such a stunt. In fact, we don’t think Sangoma would ever stand for that despite a few developers that might have a different view. But we’re warning that it’s simply not worth the risk.

Before you elect to go your own way with your own repository, be advised that importing new FreePBX-compatible GPL modules without first testing each of them with the Incredible PBX GUI is a very bad idea for the reasons already mentioned. We intend to do that with the new Incredible PBX repository, and we would encourage you to adopt the same approach.

Finally, to protect the security and integrity of your GPL Cloud resources, do not include repo.php and the contents of its accompanying src directory in your public repository. Otherwise, anyone with public access to your server would be able to change the contents of your repository. The proper methodology would be to build and maintain your repository off line and then copy the files to a public web server without the tools used to actually create and update the GPL modules and accompanying XML files. The tools themselves are GPL code, and you are more than welcome to redistribute them pursuant to the GPL license. Just don’t post them in decompressed format in your repo thereby making them functional for anonymous attacks against your repository.

To begin, download GPL-repo.tar.gz from SourceForge and decompress the tarball into a folder on your private server:

mkdir repo
cd repo
touch index.html
wget -O GPL-repo.tar.gz http://sourceforge.net/projects/pbxinaflash/files/IncrediblePBX11.11%2B11.12%20with%20Incredible%20GUI/GPL-repo.tar.gz/download
tar zxvf GPL-repo.tar.gz
yum -y install php-simplexml

The file structure will look like this where modules and src are subdirectories:

Within the modules subdirectory will be a packages subdirectory that includes folders for each of the GPL modules. There’s also a licenses folder with all of the applicable GPL licenses.

Within each of the package directories, you will find one or two modules for the two currently supported GPL versions. For example, here are the entries for the framework module:

The lists of the available modules for each supported GPL version are contained in the .xml files in the top level directory: modules-2.11.xml and all-12.0.xml. modules-12.0.xml is a symlink to a previous nomenclature for version 12. These XML files are what Module Admin uses to check for updates available for existing modules on your PBX.

To add or update individual modules in your repository, issue one or both of the following commands using the actual name of the module you wish to add or update. You can decipher the actual names for the modules by checking the FreePBX source listings on GitHub. As we cautioned previously, don’t ever add or update modules without first testing the new module on an Incredible PBX server running the Incredible GUI. If an updated module blows things up, please let us know!

./repo.php 2.11 modulename
./repo.php 12.0 modulename

And here’s how to add any compatible module from any FreePBX 2.11 or 12 server or from GitHub to your repo. On the FreePBX platform, switch to the directory holding the modules: cd /var/www/html/admin/modules. By way of example, let’s assume there’s a javassh module directory.

1. Decipher the current version of the module: grep version javassh/module.xml
2. Create a gzipped tarball of that module including the version: tar -cvzf javassh-VERSION.tgz javassh/
3. Move javassh-VERSION.tgz to your /repo folder: mv javassh-VERSION.tgz /var/www/html/repo

Alternatively, you can use the included git-grab12 script to download the latest version 12 modules in tarball format directly from the FreePBX repository on GitHub:

From your /repo folder: ./git-grab12 modulename (there is no javassh version 12 module)

4. Assimilate the javassh module into your repo as either a 2.11 or 12.0 module or both:

cd /var/www/html/repo
./repo.php 2.11 javassh-VERSION.tgz
./repo.php 12.0 javassh-VERSION.tgz
rm -f javassh-VERSION.tgz

When you’re ready to go public, move the /repo folder and its subdirectories from your private server to a public web server, issue the following commands within the main destination directory on the public server to remove the GPL repo toolkit:

rm -f git-grab12
rm -f repo.php
rm -rf src

The final step is to tell the Incredible PBX GUI the new location of your module repository. For this, you will need a fully-qualified domain name (FQDN) that points to the top-level directory of the repository stored on your public web server, e.g. http://myrepo.me.com. Once you have set up a DNS entry for this address and tested it to be sure it works, all you have to do is configure the GUI to find it. Issue the following command from the Linux CLI after logging into your server as root. Be sure to substitute your actual FQDN and your actual root password for MySQL if you have changed it from passw0rd. If you’re building a number of new servers, you could simply add this line to the end of the Incredible PBX install script. Be sure to copy the entire line below. It should end with double quotes.

mysql -u root -ppassw0rd asterisk -e "update freepbx_settings set value='http://myrepo.me.com' where keyword='MODULE_REPO' and description='repo server' limit 1"

Isn’t it amazing what you can do with some GPL code and a little documentation on how to use it? Freedom At Last!

Originally published: Tuesday, May 26, 2015


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



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


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


Some Recent Nerd Vittles Articles of Interest…

  1. According to a recent tweet from one of the developers, these warnings now can be disabled. That change was more than a year in coming. []
  2. The latest versions of the GPL modules are available in FreePBX’s GIT repo. UPDATE: Although tarballs are available for individual modules, even that format on GitHub would require painstaking, individual imports within the FreePBX GUI and totally defeats the design and purpose of the Module Admin component of FreePBX. []

Wear Something Green for May Day: The Schmoozification of Sangoma

For anyone that wants to run FreePBX® 12 with a module not produced or sold by Sangoma without being bombarded with daily emails and nasty security warnings in your GUI, here’s a portion of the agreement Sangoma would like you to sign:



And the response to those that dare claim such a practice is damaging the fabric of the Asterisk open source community:

Ward believes that he can run around, signing modules that attack other peoples machines, and then when we get sued for it he can sit back and laugh. — xrobau a.k.a. Rob Thomas, Sangoma

It’s been four months since Sangoma purchased Schmooze Com, Inc. and FreePBX®. Happy Anniversary! Silly us, thinking Sangoma was going to clean up the FreePBX mess. Here are unedited excerpts from the horse’s [insert favorite orifice] during the Sangoma free-for-all on Reddit yesterday. Read and weep…

[NOTE: What follows is data from a live feed on Reddit. For those unfamiliar with the platform, users’ comments get elevated or demoted based upon votes from other users although voting down a comment is supposed to be based upon relevance according to Reddit’s rules. User’s comments also can be edited long after the fact. Suffice it to say, there was a concerted effort to up-vote certain posts and down-vote posts that were critical of a certain point of view yesterday. In anticipation of the possibility that some comments might be physically altered in order to cast the author in a more favorable light after we published this article, we have captured all of the original text at the time this article was published. Should there be material changes in particular comments, we will post the original text below the current version so that you can draw your own conclusions.]

Original comment read as follows:

Originally published: Friday, May 1, 2015


Some Recent Nerd Vittles Articles of Interest…

Firewalls and Internet Security: Separating FUD and Fiction in the VoIP World

Some of us have spent years developing secure VoIP solutions for Asterisk® that protect your phone bill while bringing Cloud-based solutions within reach of virtually anyone. So it’s particularly disappointing when a hardware manufacturer spreads fear, uncertainty, and doubt in order to peddle their hardware. In this case, it happens to be Session Border Controllers (SBCs). We want you to watch this latest “infomercial” for yourself:



To hear Sangoma tell it, every VoIP server protected by merely a firewall is vulnerable to endless SIP attacks unless, of course, you purchase an SBC. And since implementation of Cloud-based servers traditionally limits the ability to deploy an SBC, most Cloud-based VoIP solutions would become vulnerable to SIP attacks. In the words of Sangoma:

And with telecom fraud and PBX hacking on the rise, it’s important to keep your network secure. For most enterprises, it’s not a matter of if-but-when their [sic] network experiences an attack, potentially costing you valuable time and money.

Now Sangoma is touting an article in a blog from the U.K. that begins with the headline “Why Firewalls are not Enough.” The purported author is Jack Eagle, who is otherwise unidentified. Not surprisingly, the owner of the blog happens to be a reseller of Sangoma hardware. Here’s what Jack Eagle suggests:

In addition, the inherent function of firewalls is to deny all unsolicited traffic. Whereby, the act of making a phone call is an unsolicited event, thus, firewalls can be counterproductive to an effective VoIP deployment by denying VoIP traffic.

For the benefit of those of you considering a VoIP deployment either locally or in the Cloud using Asterisk, let’s cut to the chase and directly address some of the FUD that’s been thrown out there.

FUD #1: Internet SIP Access Exposes Asterisk to Attack

False. What is true is that unrestricted SIP access to your server from the Internet without a properly secured firewall may expose Asterisk to attack. Perhaps it’s mere coincidence but the only major Asterisk aggregation that still installs Asterisk with an unsecured firewall and no accompanying script, tutorial, or even recommendation to properly lock it down and protect against SIP attacks happens to be from the same company that now wants you to buy a session border controller.

FUD #2: Firewalls Aren’t Designed to Protect Asterisk from SIP Attacks

False. What is true is that the base firewall installation provided in the FreePBX® Distro does not protect against any attacks. In a Cloud-based environment or with local deployments directly exposed to the Internet, that could very well spell disaster. And it has on a number of occasions. The Linux IPtables firewall is perfectly capable of insulating your Asterisk server from SIP attacks when properly configured. With PBX in a Flash and its open source Travelin’ Man 3 script, anonymous SIP access is completely eliminated. The same is true using the tools provided in the latest Elastix servers. And, Incredible PBX servers have always included a secured firewall with simple tools to manage it. Of course, with local VoIP hardware and a hardware-based firewall, any Asterisk server can be totally insulated from SIP attacks whether IPtables is deployed or not. Just don’t open any ports in your firewall and register your trunks with your SIP providers. Simple as that.

FUD #3: SIP Provider Access to Asterisk Compromises Your Firewall

False. Registering a server with SIP or IAX trunk providers is all that is required to provide secure VoIP communications. Calls can flow in and out of your Asterisk PBX without compromising your server or communications in any way. Contrary to what is depicted in the infomercial, there is no need to poke a hole in your firewall to expose SIP traffic. In fact, we know of only one SIP provider that requires firewall changes in order to use their services. Simple answer: use a different provider. Consider how you access Internet sites with a browser from behind a firewall. The connection from your browser to web sites on the Internet can be totally secure without any port exposure in your firewall configuration. Registering a SIP trunk with a SIP provider accomplishes much the same thing. All modern firewalls and routers will automatically handle the opening and closing of ports to accommodate the SIP or IAX communications traffic.

FUD #4: Remote Users Can’t Access Asterisk Without SIP Exposure

False. Over the past several years, we have written about a number of methodologies which allow remote users to securely access an Asterisk server. That’s what Virtual Private Networks and Port Knocking and Remote Firewall Management are all about. All of these solutions provide access without exposing your server to any SIP vulnerabilities! We hope the authors of this infomercial will give these open source tools a careful look before tarnishing the VoIP brand by suggesting vulnerabilities which any prudent VoIP deployment can easily avoid without additional cost. Just use the right products!

Originally published: Thursday, April 23, 2015



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


 
New Vitelity Special. Vitelity has generously offered a new discount for Incredible PBX 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 Incredible PBX 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…

The Gotcha-Free PBX: Simon Telephonics New SIP Gateway for Google Voice

We promised you that free Google Voice calling in the U.S. and Canada would soon be available on every Asterisk® platform whether the platform supported Asterisk Motif or not. And this week we’re covering the second SIP gateway offering for Google Voice. We introduced Bill Simon’s first Google Voice gateway back in June of 2012. This time around the latest iteration features secure OAUTH authentication so there’s no need to divulge your Google Voice credentials. Once you’ve set up your account on the Simonics Google Voice Gateway site,1 you simply create a standard SIP trunk on your Asterisk server or SIP device of choice, and PRESTO! You get secure authentication to Google Voice without worrying whether Google will drop support for insecure authentication methods such as Asterisk Motif down the road. And you can set all of it up for a one-time setup fee. For Nerd Vittles readers, you get $1 off the current $5.99 fee by using this link. Unlike last week’s GVsip offering, the new Simonics service includes free CallerID name lookups plus the ability to connect multiple devices at multiple sites and communicate between the devices using some clever SIP magic. You also can map incoming calls to any SIP URI rather than just the destination from which you register a Google Voice account. This new gateway is a real winner!

Why do this? There are several reasons aside from the free calls and free phone number. First, Google has warned for years that insecure authentication to Google Voice is going away. It hasn’t yet which is the reason Asterisk Motif logins still work. When Google finally pulls the plug (and they will), your Google Voice days are over using the Asterisk platform. Second, some of the Asterisk aggregations such as Elastix® never supported Google Motif. Hence, free Google Voice calling wasn’t available at all to those using the Elastix platform. That limitation is now a thing of the past. You can create a simple SIP trunk and begin enjoying free Google Voice calling in the U.S. and Canada just like some of the rest of us have been doing for years. Third, Google Voice support was the sole reason that many have stuck with the FreePBX® GUI despite the gotchas. Now you have a choice. Any Incredible PBX™ or Asterisk-GUI™ server now supports Google Voice without your having to worry about constant changes to the Asterisk Motif driver to support refinements at the Google Voice end. Now it’s a pure SIP trunk using pure SIP technology as far as Asterisk is concerned. The only limitation is the one imposed by Google. You need to reside in the United States to use Google Voice even though free calling is available to the U.S. and Canada.

If you have difficulty finding the Google Chat option after setting up a new Google Voice account, follow this tutorial.

1. Using your favorite browser, log in to the Google Voice account you wish to associate with the Simonics SIP gateway. Be sure that you’ve enabled Google Chat in your Google Voice setup.

2. Using a separate tab of your browser, connect to the Simonics Google Voice Gateway site.

3. Go through the steps to register your Google Voice account with the Simonics Google Voice gateway and obtain your credentials.

4a. For those using FreePBX or Elastix, use another tab of your browser to open the GUI interface and create a new SIP trunk using your new SIP login credentials. Replace 8005551212 with your actual Google Voice number and YOUR-SIP-PW with your actual Simonics SIP password in BOTH the PEER Details and Registration String. Add your Google Voice number to the end of the Registration String like this: GV18005551212:YOUR-SIP-PW@gvgw.simonics.com/8005551212

4b. For those using Incredible PBX for Asterisk-GUI, simply download and run our One-Click Installer. You’ll need your Simonics SIP account name and password plus a two-digit dialing prefix to use for outbound calls. It’s that simple!

cd /root
wget http://incrediblepbx.com/simonics-addon.tar.gz
tar zxvf simonics-addon.tar.gz
rm -f simonics-addon.tar.gz
./simonics-addon.sh

Once you’ve finished running the script, your trunk will be up and running. There’s no requirement for steps #5 and #6 with Asterisk-GUI. If desired, jump to Step #7 to set up a SIP URI for your incoming calls.

5. Create an Inbound Route for your incoming calls using the 10-digit number you entered at the end of the Registration String in step #4a.

6. Create an Outbound Route for outgoing calls that should be handled by your Google Voice trunk. The CallerID number will be your Google Voice number. You cannot change it.

7. If you’d prefer to send incoming calls to a designated SIP URI instead of the server that registered with the Simonics gateway, enter the address in the format: pbx@myserver.xyz. For additional details, read our previous article on SIP URIs.

8. Repeat this setup procedure for as many Google Voice accounts as you wish to activate using the steps above. If you’re using Incredible PBX for Asterisk-GUI, remember to edit the script and change the TRUNK=simonics entry to something like TRUNK=simonics2. Also use a unique two-digit dialing prefix for each trunk. Be sure to logout of your previous Google account before repeating the drill. Enjoy!


Don’t forget to List Yourself in Directory Assistance with your new IPkall PSTN number so everyone can find you by dialing 411. And be sure to add your new number to the Do Not Call Registry to block telemarketing calls.

Originally published: Monday, April 13, 2015


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



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


 
New Vitelity Special. Vitelity has generously offered a new discount for Incredible PBX 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 Incredible PBX 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. In addition to substantial technical assistance, Simon Telephonics is also a financial contributor to the Nerd Vittles project. []

The Two Amigos on Cloud 9: Introducing Incredible PBX for Elastix @ RentPBX

We continue the Gotcha-Free PBX adventure today with an open source alternative for which many have been clamoring, another affordable Cloud-based Asterisk® platform with the no-strings-attached Elastix 2.5 GUI. In addition to a $15 a month hosting plan, the icing on the cake is the quick 10-minute automated setup on your choice of a dozen servers throughout the U.S. as well as Canada and Europe. If you can find the Enter key on a keyboard, then you can handle the complexity of the RentPBX setup for Incredible PBX for Elastix 2.5. When you’re finished, you’ll have a turnkey PBX featuring some terrific open source software. The software is all free, subject only to the terms of the open source licenses.

Target Audience: Home or Office in need of a turnkey, Gotcha-Free Elastix PBX in the Cloud

Default Configuration: Asterisk 11 with enhanced Elastix 2.5 GUI

Platform: CentOS 5.11 running on RentPBX Cloud-Based Server platform

Memory: 400 MB with 415 MB swap

Disk Size: 20 GB

Default Trunks: CallCentric, DIDlogic, Future-Nine, IPcomms, Les.net, Vitelity, VoIP.ms, Gvoice1

Feature Set: Fax, SMS messaging, NeoRouter/PPTP VPN, Reminders, ConfBridge Conferencing, AsteriDex, Voicemail, Email, IVR, News, Weather, Voice Dialer, Wolfram Alpha, Today in History, TM3 Firewall WhiteList, Speed Dialer, iNUM and SIP URI (free) worldwide calling, DISA, Call Forwarding, Tailorable CDRs

Administrator Utilities: Incredible Backup/Restore, Automatic Updater, phpMyAdmin, Timezone Config, WebMin, Admin Password Configurator, ODBC/MySQL Database Configurator, Firewall WhiteList Tools

Getting Started with Incredible PBX for Elastix 2.5 (Cloud Edition)

Here’s a quick overview of the installation and setup process for Incredible PBX for Elastix 2.5 @ RentPBX.com:

  1. Sign Up for Incredible PBX for Elastix 2.5 in the Cloud
  2. Complete the Install of Incredible PBX with two automatic reboots
  3. Set Up Passwords for Incredible PBX
  4. Configure Trunks with Incredible PBX
  5. Connect a Softphone to Incredible PBX
  6. Configure SMTP Mail for Incredible PBX

1. Sign Up for Incredible PBX for Elastix 2.5 in the Cloud at RentPBX.com

Visit RentPBX.com and choose the Elastix build option. Then complete the following steps:

Step #1. Select a location for your cloud-based server.

Step #2. Choose Elastix 2.5 IncrediblePBX Ready option.

Step #3. Specify a hostname for your server.

Step #4. When you begin the payment/checkout phase, enter your coupon code to take advantage of the $15/month discounted rate: NOGOTCHAS. Wait for the confirmation email with your server credentials and dedicated IP address.

2. Complete the Install of Incredible PBX

Nothing tricky here. It’s a 10-minute automated setup. Log into port 20022 of your server as root with your default password using SSH or Putty. Once you’re logged in, RentPBX will go through two setup cycles to complete the install and randomize all of your passwords for Incredible PBX. The first pass addresses some security vulnerabilities in the Elastix 2.5 base install and then prompts for the MySQL root password which must be passw0rd (with a zero). Next, you’re prompted to set up an admin password for the GUI. Make it secure! Then your server will reboot. After 60 seconds, log back in to port 20022 as root with your default password again. Type y to install Incredible PBX. Incredible PBX will first apply the latest upgrades for CentOS and Elastix. Be patient. The list is a long one. After the second reboot, log back into your server on port 20022 as root one final time and let Incredible PBX complete the install and secure your server. You’ll need to enter your MySQL and GUI passwords once again. Be sure to use passw0rd for MySQL! After the third reboot, log back into your server on the standard port 22 as root. Allow Incredible PBX to run its Automatic Update Utility to bring your system current. That’s it. You now have a secure, turnkey Elastix® PBX that’s ready for use.

3. Initial Configuration of Incredible PBX for Elastix 2.5

Incredible PBX is installed with the preconfigured IPtables Linux firewall already in place. It implements WhiteList Security to limit server access to your server’s IP address, your desktop computer’s IP address, and a few of our favorite SIP providers. You can add additional entries to this WhiteList whenever you like using the add-ip and add-fqdn tools in /root. There’s also an Apache security layer for web applications. And, of course, Elastix 2.5 has its own security methodology. RentPBX randomized extension and DISA passwords as part of the initial setup process. Out of the starting gate, you won’t find a more secure VoIP server implementation anywhere. After all, it’s your phone bill.

Even with all of these layers of security, here are 5 Quick Steps to better safeguard your server. You only do this once, but failing to do it may lead to security issues you don’t want to have to deal with down the road. So DO IT NOW!

Log into your server as root with your root password and do the following:

Make your root password very secure: passwd
Set your correct time zone: ./timezone-setup
Create admin password for web apps: htpasswd -b /etc/pbx/wwwpasswd admin newpassword
Make a copy of your other passwords: cat passwords.FAQ
Decipher IP address and other info about your server: status

Using a browser, you’re not ready to log into the Elastix 2.5 GUI with your new admin password.

4. Activate Trunks with Incredible PBX for Elastix 2.5

For those migrating from another aggregation including PBX in a Flash, this should be familiar territory for you. Using a browser, log into Elastix 2.5 at the IP address of your server. Before you can actually make or receive calls outside your PBX, you’ll need at least one trunk. In the Elastix 2.5 GUI, click PBX -> Trunks. Once you have your credentials from a provider, choose a provider from the list of preconfigured trunks on the right or create a new one. If you’re using one of the preconfigured options, remember to enable the trunk after adding your desired CallerID and credentials. Then save your settings and reload your Asterisk dialplan. That’s it. You’re ready to go.

5. Configure a Softphone with Incredible PBX for Elastix 2.5

Incredible PBX comes preconfigured with two extensions (701 and 702) that let you connect phones to your PBX. You can connect virtually any kind of telephone to your Elastix 2.5 PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. You can find them in /root/passwords.FAQ. Fill in the blanks using the IP address of your server, 701 for your account name, and whatever password is assigned to the extension. Here’s what your entries should look like. Click OK to save your entries.

Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Here are a few numbers to get you started:


123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
222 - ODBC Lookup (try: 12345)
DEMO - Allison's IVR Demo
TODAY - Today in History

6. Configuring SMTP Mail with Incredible PBX for Elastix 2.5

Outbound email support using Postfix is preconfigured with Elastix 2.5. You can test whether it’s actually working by issuing the following command using your destination email address after logging in as root:

echo "test" | mail -s testmessage yourname@gmail.com

If you don’t receive the email message within a minute or two and you’ve checked your spam folder, chances are your ISP is blocking downstream SMTP servers in an effort to combat spam. Comcast is one of the usual suspects. To enable outbound email service for delivery of voicemail and other email messages with a provider blocking downstream SMTP servers, you first need to obtain the SMTP domain of your ISP, e.g. smtp.comcrap.net. Next, edit /etc/postfix/main.cf and add your SmartHost entry [in brackets] to the line that begins like this: relayhost =. The line should look like this: relayhost = [smtp.comcrap.net]. Save your addition and restart Postfix: service postfix restart. Be sure to try another email test message after completing the SmartHost update. To use Gmail as your mail relay, see this tutorial.

Configuring Google Voice

We have included the Python implementation of gvoice in /root for those that want to experiment by making calls and sending SMS blasts the “old-fashioned” way. While Elastix does not directly support native Asterisk 11 Google Voice functionality, you now can use a SIP gateway to access Google Voice and make free calls in the U.S. and Canada.

Homework Assignment: Mastering Incredible PBX for Elastix 2.5

We’ve put together a complete tutorial for the applications included in Incredible PBX for Asterisk-GUI. Most of it is fully applicable to Elastix 2.5 as well. That should be your next stop. Then you’ll be ready to tackle Elastix 2.5. Google is your friend. Do some exploring, and we’ll post links to great articles on this terrific platform as we discover them. Your suggestions are also welcomed!

In the meantime, if you have questions, join the PBX in a Flash Forums and take advantage of our awesome collection of gurus. There’s an expert available on virtually any topic, and the price is right. As with Incredible PBX, it’s absolutely free. The same applies to the Elastix forum.

And if all of that wasn’t enough, feast your eyes on the Elastix Add-Ons that are only a button click away:

Download (PDF, 619KB)

Originally published: Friday, March 27, 2015


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



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


 
New Vitelity Special. Vitelity has generously offered a new discount for Incredible PBX 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 Incredible PBX 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. Vitelity, Google, and RentPBX provide financial support to Nerd Vittles and the Incredible PBX project. []

The Two Amigos Ride Again: Introducing Incredible PBX for Elastix 2.5

We began our Elastix® adventure last week with the Bleeding Edge and our introduction of Incredible PBX for Elastix MT, the promising new multi-tenant edition of Elastix. Unfortunately, for production use, Elastix 3.0 is not quite there yet. So this week we’re introducing Incredible PBX™ for Elastix 2.5, an incredibly stable telephony platform with a loyal following and dozens of add-on components to satisfy almost any requirement. Having not looked at Elastix in more than a year, we were pleasantly surprised to find a very current version of Asterisk® 11 as well as a stable, gotcha-free Elastix fork of FreePBX® 2.11. It’s amazing what can be accomplished with a single command: yum upgrade. If you know how to use FreePBX, then the Elastix GUI will be a walk in the park.

We promised that 2015 would be the year of Gotcha-Free choices for the Asterisk platform, and today we deliver the third VoIP alternative with pure and honest GPL code minus the patent, trademark, and copyright minefields previously covered. Incredible PBX™ for Elastix 2.5 provides virtually the same feature set of applications for Asterisk as our previous releases. Just abide by the clear GPL licensing terms and copy, embellish, and redistribute to your heart’s content.

What Incredible PBX brings to the Elastix 2.5 platform are several dozen (free) applications for Asterisk in addition to a rock-solid firewall with a preconfigured WhiteList of your favorite VoIP providers and private LAN addresses. With the Elastix 2.5 version, you also get a dozen preconfigured trunks and extensions plus a familiar GUI that we’ve all used for the better part of a decade. And it’s all bundled in a graphical user interface that integrates telephony, faxing, instant messaging, email, and calendaring in a single desktop application. We’re glad to be part of the family.

Our deployment strategy remains consistent and straight-forward. Install a 64-bit bit version of Elastix 2.5 on the platform of your choice. Then run the Incredible PBX installer. In 5-10 minutes, you’re ready to roll. The installer first will bring Elastix 2.5 and CentOS up to current specs. Then it will work its magic and add an Incredible PBX tab to the existing Elastix 2.5 UI with all the bells and whistles to which you are accustomed. Text-to-speech applications, speech recognition, DISA, ODBC, SMS messaging, news, weather, conference bridge support, and a voice dialer are enabled out of the box.

A Word of Caution. If you’re new to Incredible PBX, install a clean version of Elastix 2.5 with NO MODIFICATIONS before you begin the Incredible PBX install. All of the existing Elastix 2.5 setup will be modified as part of the Incredible PBX install, and these changes will wipe out any additions you’ve previously made to Elastix. So don’t make any! Once the Incredible PBX install is completed, you can make all the changes you wish in your Elastix configuration. The only major design change we’ve made is to rework the Elastix MySQL database tables into MyISAM format from InnoDB. This facilitates making future backups and restores of your server as well as providing the necessary platform to install current and future Incredible PBX components.

Did We Mention Security? You also get a locked down, preconfigured IPtables Firewall WhiteList with all of the Travelin’ Man 3 tools plus the automatic update service to keep your server up to date and safe. There is a $20 voluntary annual license fee for the update service but, if you’d prefer to buy donuts, be our guest. But understand that voluntary is a two-way street. Running the update service costs us time and money and, when it ceases to be worthy of our time and financial investment, we reserve the right to discontinue the service down the road. The next time you log into your server after installing Incredible PBX, you’ll quickly appreciate why an automatic update service is important. We watch for and fix problems so you don’t have to.

Target Audience: Small or Large Organization in need of a turnkey, Gotcha-Free PBX

Default Configuration: Asterisk 11 with enhanced Elastix 2.5 GUI and Kennonsoft GUI

Platform: CentOS 5.x running on Dedicated Server, Cloud-Based Server, or Virtual Machine

Minimum Memory: 1024 MB

Recommended Disk: 20 GB+

Feature Set: Fax, SMS messaging, NeoRouter/PPTP VPN, Reminders, ConfBridge Conferencing, AsteriDex, Voicemail, Email, IVR, News, Weather, Voice Dialer, Wolfram Alpha, Today in History, TM3 Firewall WhiteList, Speed Dialer, iNUM and SIP URI (free) worldwide calling, DISA, Call Forwarding, Tailorable CDRs

Administrator Utilities: Incredible Backup/Restore, Automatic Updater, phpMyAdmin, Timezone Config, WebMin, Admin Password Configurator, ODBC/MySQL Database Configurator, Firewall WhiteList Tools

Getting Started with Incredible PBX for Elastix 2.5

Here’s a quick overview of the installation and setup process for Incredible PBX for Elastix 2.5:

  1. Choose a Hardware Platform – Dedicated PC, Cloud Provider, or Virtual Machine
  2. Install Elastix 2.5 – 64-bit CentOS 5 platform
  3. Download and Install Incredible PBX for Elastix 2.5
  4. Set Up Passwords for Incredible PBX for Elastix 2.5
  5. Activate Trunks with Incredible PBX for Elastix 2.5
  6. Connect a Softphone to Incredible PBX for Elastix 2.5
  7. Configuring SMTP Mail with Incredible PBX for Elastix 2.5

1. Choose a Platform for Incredible PBX for Elastix 2.5

Incredible PBX for Elastix 2.5 works equally well on dedicated hardware, a cloud-based server, or a virtual machine. Just be sure you’ve met the minimum requirements outlined above and that you have a sufficiently robust Internet connection to support 100Kb of download and upload bandwidth for each simultaneous call you wish to handle with your new PBX.

For Dedicated Hardware, we recommend at least an Atom-based PC of recent vintage with at least a 30GB drive and 4GB of RAM. That will take care of an office with 10-20 extensions and a half dozen or more simultaneous calls if you have the Internet bandwidth to support it.

For Cloud-Based Servers, we recommend RentPBX, one of our financial supporters who also happens to size servers properly and restrict usage solely to VoIP. This avoids performance bottlenecks that cause problems with VoIP calls. Yes, we have a coupon code for you to get the $15/month rate: NOGOTCHAS. The new image to support Incredible PBX for Elastix 2.5 should be available shortly.

For Virtual Machine Installs, we recommend Oracle’s VirtualBox platform which runs atop almost any operating system including Windows, Macs, Linux, and Solaris. Here’s a link to our original VirtualBox tutorial to get you started. We suggest allocating 1GB of RAM and at least a 20GB disk image to your virtual machine for best performance. We actually used VirtualBox to build Incredible PBX for Elastix 2.5.

2. Install 64-bit Elastix 2.5 on Your Platform

Begin by downloading the 64-bit Elastix 2.5 ISO. For dedicated hardware, burn the ISO image to a CD/DVD and boot your server with the Elastix 2.5 ISO to begin the install. Here are the simplest installation steps:

Install or Upgrade in Graphical Mode by pressing ENTER
Choose: Install Language
Choose: Keyboard
Choose: Initialize Drive and Erase ALL DATA
Remove: All partitions on selected drive and YES you’re sure
Modify: Partitioning Layout (No is fine)
Configure: eth0 and disable IPv6 Support (unless required)
Choose: Dynamic IP (DHCP) configuration
Choose: Hostname Configuration Automatic
Choose: Time Zone and Disable System Clock Uses UTC
Set: Root Password (Make it Secure!)
Wait for Reboot to Complete
Set MySQL Password to: passw0rd (MANDATORY: with a zero!)
Choose Elastix admin Password: minimum 10 alphanumeric characters with upper & lowercase

For VirtualBox, create an Elastix 2.5 virtual machine of Linux (RedHat 64-bit) type by clicking New. Click Settings button. In System, enable I/O APIC and disable Hardware Clock in UTC Time. In Audio, enable Audio for your sound card. In Network, enable Bridged Adapter for Adapter 1. In Storage, click on Empty in the Storage Tree. Then click on the Disk icon to the right of CD/DVD Drive attributes. Choose the Elastix 2.5 ISO file that you downloaded. Click OK. Then start the virtual machine to begin the installation process. Follow the setup steps above to install Elastix 2.5 in your virtual machine.

3. Download and Install Incredible PBX for Elastix 2.5

After completing the Elastix 2.5 install, log into your server as root using SSH or Putty from a desktop machine that you will use to manage your server. This is important with the Incredible PBX IPtables Firewall WhiteList so you don’t get locked out of your own server! Then issue the following commands to begin the Incredible PBX install. You’ll actually run the installer twice, once to upgrade CentOS and Elastix and a second time to install Incredible PBX.

cd /root
wget http://incrediblepbx.com/incrediblepbx11elastix25.tar.gz
tar zxvf incrediblepbx11elastix25.tar.gz
rm -f incrediblepbx11elastix25.tar.gz
./IncrediblePBX11-Elastix25.sh
./IncrediblePBX11-Elastix25.sh

4. Initial Configuration of Incredible PBX for Elastix 2.5

Incredible PBX is installed with the preconfigured IPtables Linux firewall already in place. It implements WhiteList Security to limit server access to connected LANs, your server’s IP address, your desktop computer’s IP address, and a few of our favorite SIP providers. You can add additional entries to this WhiteList whenever you like using the add-ip and add-fqdn tools in /root. There’s also an Apache security layer for our web applications. And, of course, Elastix 2.5 has its own security methodology. Finally, we randomize extension and DISA passwords as part of the initial install process. Out of the starting gate, you won’t find a more secure VoIP server implementation anywhere. After all, it’s your phone bill.

Even with all of these layers of security, here are 6 Quick Steps to better safeguard your server. You only do this once, but failing to do it may lead to security issues you don’t want to have to deal with down the road. So DO IT NOW!

First, log out and back into your server as root with your root password to get the latest updates. Then do the following:

Make your root password very secure: passwd
Set your correct time zone: ./timezone-setup
Create admin password for web apps: htpasswd -b /etc/pbx/wwwpasswd admin newpassword
Set MySQL and Elastix admin PW: ./admin-pw-change (MySQL PW MUST be passw0rd with zero)
Make a copy of your other passwords: cat passwords.FAQ
Decipher IP address and other info about your server: status

Last but not least, Incredible PBX includes an automatic update utility which downloads important updates whenever you log into your server as root. We recommend you log in once a week to keep your server current. If you haven’t already done so, NOW would be a good time to log out and back into your server at the Linux command line to bring your server current.

5. Activate Trunks with Incredible PBX for Elastix 2.5

For those migrating from another aggregation including PBX in a Flash, this should be familiar territory for you. Using a browser, log into Elastix 2.5 at the IP address of your server. Before you can actually make or receive calls outside your PBX, you’ll need at least one trunk. In the Elastix 2.5 GUI, click PBX -> Trunks. Once you have your credentials from a provider, choose a provider from the list of preconfigured trunks on the right or create a new one. If you’re using one of the preconfigured options, remember to enable the trunk after adding your desired CallerID and credentials. Then save your settings and reload your Asterisk dialplan. That’s it. You’re ready to go.

6. Configure a Softphone with Incredible PBX for Elastix 2.5

Incredible PBX comes preconfigured with two extensions (701 and 702) that let you connect phones to your PBX. You can connect virtually any kind of telephone to your Elastix 2.5 PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. You can find them in /root/passwords.FAQ. Fill in the blanks using the IP address of your server, 701 for your account name, and whatever password is assigned to the extension. Here’s what your entries should look like. Click OK to save your entries.

Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Here are a few numbers to get you started:


123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
222 - ODBC Lookup (try: 12345)
DEMO - Allison's IVR Demo
TODAY - Today in History

6. Configuring SMTP Mail with Incredible PBX for Elastix 2.5

Outbound email support using Postfix is preconfigured with Elastix 2.5. You can test whether it’s actually working by issuing the following command using your destination email address after logging in as root:

echo "test" | mail -s testmessage yourname@gmail.com

If you don’t receive the email message within a minute or two and you’ve checked your spam folder, chances are your ISP is blocking downstream SMTP servers in an effort to combat spam. Comcast is one of the usual suspects. To enable outbound email service for delivery of voicemail and other email messages with a provider blocking downstream SMTP servers, you first need to obtain the SMTP domain of your ISP, e.g. smtp.comcrap.net. Next, edit /etc/postfix/main.cf and add your SmartHost entry [in brackets] to the line that begins like this: relayhost =. The line should look like this: relayhost = [smtp.comcrap.net]. Save your addition and restart Postfix: service postfix restart. Be sure to try another email test message after completing the SmartHost update. To use Gmail as your mail relay, see this tutorial.

Configuring Google Voice

We have included the Python implementation of gvoice in /root for those that want to experiment by making calls and sending SMS blasts the “old-fashioned” way. While Elastix does not directly support native Asterisk 11 Google Voice functionality, you now can use a SIP gateway to access Google Voice and make free calls in the U.S. and Canada.

If you have difficulty finding the Google Chat option after setting up a new Google Voice account, follow this tutorial.

Homework Assignment: Mastering Incredible PBX for Elastix 2.5

We’ve put together a complete tutorial for the applications included in Incredible PBX for Asterisk-GUI. Most of it is fully applicable to Elastix 2.5 as well. That should be your next stop. Then you’ll be ready to tackle Elastix 2.5. Google is your friend. Do some exploring, and we’ll post links to great articles on this terrific platform as we discover them. Your suggestions are also welcomed!

In the meantime, if you have questions, join the PBX in a Flash Forums and take advantage of our awesome collection of gurus. There’s an expert available on virtually any topic, and the price is right. As with Incredible PBX, it’s absolutely free. The same applies to the Elastix forum.

And if all of that wasn’t enough, feast your eyes on the Elastix Add-Ons that are only a button click away:

Download (PDF, 619KB)

Originally published: Tuesday, March 10, 2015


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



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


 
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