Posts tagged: freepbx

Introducing Incredible PBX 11-12 with Incredible GUI for the Ubuntu 14 Platform

On May 15, we turned the page on Asterisk® GUIs by introducing a new GUI that hopefully provides the best of both worlds. It preserves the GPL components of the FreePBX® product that many of us have nurtured for almost a decade while removing the commercial pieces that have introduced some friction into the equation for users and companies that simply wished to deploy or redistribute a graphical user interface for Asterisk in accordance with the free GPL licenses under which the product and its components were licensed. Last week we did much the same thing with the essential cloud component which serves as the lynchpin for GPL module administration within the GUI itself. Hopefully, these two tweaks will encourage Sangoma, the new owner of the FreePBX project, to do the right thing and get the non-commercial pieces of the project back on the right track moving forward. What we did not want to do was tarnish the incredibly hard work that dozens of developers in the open source community have poured into this project over the past decade. We continue to be amazed at what they’ve been able to achieve, and we salute their accomplishments. The Asterisk 12 and 13 revolution never would have been achieved without the contributions of the FreePBX development team. We think the new Incredible PBX GUI stands as a testament to what can be accomplished while preserving the true spirit of open source development and the terms of the GPL licenses under which this product and its numerous modules were licensed.

Two weeks ago, we introduced the all-new Incredible PBX with Incredible GUI for CentOS, Scientific Linux, and Oracle Linux. Last week we added a Cloud-based GPL repository and all the tools necessary to maintain it. Today we’re pleased to release the production-ready version for the Ubuntu 14 platform with all the bells and whistles including Incredible Fax featuring HylaFax and AvantFax. Today’s release mimics the functionality of the previous build for the CentOS platform with literally dozens of turnkey applications that show off the very best features of Asterisk®. In addition to Incredible PBX, you also get our new GPL repository to maintain release 12 of the GUI. No strings, no gotchas, and no murky licenses. Pure GPL!

Building an Ubuntu 14.04 Platform for Incredible PBX

As a result of the trademark and copyright morass, we’ve steered away from the bundled operating system in favor of a methodology that relies upon you to put in place the operating system platform on which to run PBX in a Flash or Incredible PBX. The good news is it’s easy! With many cloud-based providers1, you can simply click a button to choose your favorite OS flavor and within minutes, you’re ready to go. With many virtual machine platforms such as VirtualBox, it’s equally simple to find a pre-built Ubuntu 14.04 image or roll your own.

If you’re new to VoIP or to Nerd Vittles, here’s our best piece of advice. Don’t take our word for anything! Try it for yourself in the Cloud! You can build an Ubuntu 14.04 image on Digital Ocean in under one minute and install today’s Incredible PBX for Ubuntu 14.04 in about 15 minutes. Then try it out for two full months. It won’t cost you a dime. Use our referral link to sign up for an account. Enter a valid credit card to verify you’re who you say you are. Create an Ubuntu 14.04 (not 14.10!) 512MB droplet of the cheapest flavor ($5/mo.). Go to the Billing section of the site, and enter the following promo code: UBUNTUDROPLET. That’s all there is to it. A $10 credit will be added to your account, and you can play to your heart’s content. Delete droplets, add droplets, and enjoy the free ride!

For today, we’ll walk you through building your own stand-alone server using the Ubuntu 14.04 mini.iso. If you’re using Digital Ocean in the Cloud, skip down to Installing Incredible PBX 11-12 (HINT: 11 tells you the Asterisk release and 12 tells you the GUI release). If you’re using your own hardware, to get started, download the 64-bit Ubuntu 14.04 “Trusty Tahr” Minimal ISO from here. Then burn it to a CD/DVD or thumb drive and boot your dedicated server from the image. Remember, you’ll be reformatting the drive in your server so pick a machine you don’t need for other purposes.

For those that would prefer to build your Ubuntu 14.04 Wonder Machine using VirtualBox on any Windows, Mac, or existing Linux Desktop, here are the simple steps. Create a new virtual machine specifying the 64-bit version of Ubuntu. Allocate 1024MB of RAM (512MB also works fine with a swap file) and at least 20GB of disk space using the default hard drive setup in all three steps. In Settings, click System and check Enable I/O APIC and uncheck Hardware Clock in UTC Time. Click Audio and Specify then Enable your sound card. Click Network and Enable Network Adapter for Adapter 1 and choose Bridged Adapter. Finally, in Storage, add the Ubuntu 14.04 mini.iso to your VirtualBox Storage Tree as shown below. Then click OK and start up your new virtual machine. Simple!

Here are the steps to get Ubuntu 14.04 humming on your new server or virtual machine once you’ve booted up. If you can bake cookies from a recipe, you can do this:

UBUNTU mini.iso install:
Choose language
Choose timezone
Detect keyboard
Hostname: incrediblepbx < continue >
Choose mirror for downloads
Confirm archive mirror
Leave proxy blank unless you need it
< continue >
** couple minutes of whirring as initial components are loaded **
New user name: incredible
< continue >
Account username: incredible
< continue >
Account password: makeitsecure
< continue >
Encrypt home directory < no >
Confirm time zone < yes >
Partition disks: Guided - use entire disk and set up LVM
Confirm disk to partition
Write changes to disks and configure LVM
Whole volume? < continue>
Write changes to disks < yes> < -- last chance to preserve your disk drive!
** about 15 minutes of whirring during base system install ** < no touchy anything>
** another 5 minutes of whirring during base software install ** < no touchy anything>
Upgrades? Install security updates automatically
** another 5 minutes of whirring during more software installs ** < no touchy anything>
Software selection: *Basic Ubuntu server (only!)
** another couple minutes of whirring during software installs ** < no touchy anything>
Grub boot loader: < yes>
UTC for system clock: < no>
Installation complete: < continue> after removing installation media
** on VirtualBox, PowerOff after reboot and remove [-] mini.iso from Storage Tree & restart VM
login as user: incredible
** enter user incredible's password **
sudo passwd
** enter incredible password again and then create secure root user password **
su root
** enter root password **
apt-get update
apt-get install ssh -y
sed -i 's|without-password|yes|' /etc/ssh/sshd_config
sed -i 's|yes"|without-password"|' /etc/ssh/sshd_config
sed -i 's|"quiet"|"quiet text"|' /etc/default/grub
update-grub
ifconfig
** write down the IP address of your server from ifconfig results
reboot
** login via SSH to continue **

Installing Incredible PBX 11-12 on Your Ubuntu 14.04 Server

Adding Incredible PBX 11-12 to a running Ubuntu 14.04 server is a walk in the park. To restate the obvious, your server needs a reliable Internet connection to proceed. Using SSH (or Putty on a Windows machine), log into your new server as root at the IP address you deciphered in the ifconfig step at the end of the Ubuntu install procedure above. First, make sure to run the update step for Ubuntu below before you begin the install. This is especially important if you’re using a cloud-based Ubuntu 14 server. If errors appear during the update, just run it again.

apt-get update && apt-get upgrade -y && reboot

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

Now let’s begin the Incredible PBX 11-12 install. Log back in as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/incrediblepbx11-12.1-ubuntu14.tar.gz
tar zxvf incrediblepbx*
#./create-swapfile-DO
./Incredible*

Once you have agreed to the license agreement and terms of use, press Enter and go have a 30-minute cup of coffee. The Incredible PBX installer runs unattended so find something to do for a bit unless you just like watching code compile. When you see “Have a nice day”, your installation is complete. Write down your admin password for the GUI as well as your three “knock” ports for PortKnocker. If you forget your admin password or wish to change it, just run: /root/admin-pw-change. Retrieve your PortKnocker setup like this: cat /root/knock.FAQ.

Log out and back in as root and you should be greeted with a status display that looks something like this after the Automatic Update Utility runs:

Perform the following steps:

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

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. Now would be a good time to log out and back into your server at the Linux command line to bring your server up to current specs.

You can access the Incredible PBX GUI using your favorite web browser to configure your server. Just enter the IP address shown in the status display.

When the Kennonsoft menu (shown above) appears, click on the User tab to open the Admin menu. Then click on Incredible GUI Administration to access the Incredible PBX GUI. The default username is admin with the randomized password you wrote down above. If desired, you can change them after logging into the GUI by clicking Admin -> Administrators -> admin. Enter a new password and click Submit Changes then Apply Config. Now edit extension 701 so you can figure out (or change) the randomized passwords that were set up for default 701 extension and voicemail: Applications -> Extensions -> 701.

Setting Up a Soft Phone to Use with Incredible PBX

Now you’re ready to set up a telephone so that you can play with Incredible PBX. We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of your server, 701 for your account name, and whatever password you created for the extension. Click OK.

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

123 - Reminders
222 - ODBC Demo (use acct: 12345)
947 - Weather by ZIP Code
951 - Yahoo News
*61 - Time of Day
*68 - Wakeup Call
TODAY - Today in History

The next step is establishing an interface on your PBX to connect to the telephones in the rest of the world. If you live in the U.S., the easiest way (at least for now) is to use an existing (free) Google Voice account. Google has threatened to shut this down but as this is written, it still works with previously set up Google Voice accounts. The more desirable long-term solution is to choose several SIP providers and set up redundant trunks for your incoming and outbound calls. The PIAF Forum includes dozens of recommendations to get you started.

Configuring Google Voice

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

We’ve tested this extensively using an existing Google Voice account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Use a previously configured and dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX 11.

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you’re probably out of luck. Google has disabled the option in newly created accounts as well as some old ones that had Google Chat disabled. Now go back to the Google Voice Settings.

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

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

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

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

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

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

If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

And here’s another way to access Google Voice securely using an inexpensive commercial SIP gateway:

Troubleshooting Audio and DTMF Problems

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

Adding Speech Recognition to Incredible PBX

To support many of our applications, Incredible PBX has included Google’s speech recognition service for years. These applications include Weather Reports by City (949), AsteriDex Voice Dialing by Name (411), and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for “personal and development use.” If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!

1. Using an existing Google/Gmail account to join the Chrome-Dev Group.

2. Using the same account, create a new Speech Recognition Project.

3. Click on your newly created project and choose APIs & auth.

4. Turn ON Speech API by clicking on its Status button in the far right margin.

5. Click on Credentials in APIs & auth and choose Create New Key -> Server key. Leave the IP address restriction blank!

6. Write down your new API key or copy it to the clipboard.

7. Log into your server as root and issue the following commands:

# for Ubuntu and Debian platforms
apt-get clean
apt-get install libjson-perl flac -y
# for RedHat and CentOS platforms
yum -y install perl-JSON
# for all Linux platforms
cd /var/lib/asterisk/agi-bin
mv speech-recog.agi speech-recog.last.agi
wget --no-check-certificate https://raw.githubusercontent.com/zaf/asterisk-speech-recog/master/speech-recog.agi
chown asterisk:asterisk speech*
chmod 775 speech*
nano -w speech-recog.agi

8. When the nano editor opens, go to line 70 of speech-recog.agi: my $key = "". Insert your API key from Step #6 above between the quotation marks and save the file: Ctrl-X, Y, then Enter.

Now you’re ready to try out the speech recognition apps. Dial 949 and say the name of a city and state/province/country to get a current weather forecast from Yahoo. Dial 411 and say “American Airlines” to be connected to American.

To use Wolfram Alpha by phone, you first must install it. Obtain your free Wolfram Alpha APP-ID here. Then run the one-click installer: /root/wolfram/wolframalpha-oneclick.sh. Insert your APP-ID when prompted. Now dial 4747 to access Wolfram Alpha by phone and enter your query, e.g. “What planes are overhead.” Read the Nerd Vittles tutorial for additional examples and tips.

A Few Words about the Incredible PBX Security Model for Ubuntu

Incredible PBX for Ubuntu 14 is a very secure, turnkey PBX implementation. As configured, your server is protected by both Fail2Ban and a hardened configuration of the IPtables Linux firewall. Nobody can access your PBX without your credentials AND an IP address that is either on your private network or that matches the IP address of your server or the PC from which you installed Incredible PBX. Incredible PBX is preconfigured to let you connect to many of the leading SIP hosting providers without additional firewall tweaking.

You can whitelist additional IP addresses for remote access in several ways. First, you can use the command-line utilities: /root/add-ip and /root/add-fqdn. You can also remove whitelisted IP addresses by running /root/del-acct. Second, you can dial into extension 864 (or use a DID pointed to extension 864 aka TM4) and enter an IP address to whitelist. Before Travelin’ Man 4 will work, you’ll need to add credentials for each caller using the tools in /root/tm4. You must add at least one account before dial-in whitelisting will be enabled. Third, you can temporarily whitelist an IP address by successfully executing the PortKnocker 3-knock code established for your server. You’ll find the details and the codes in /root/knock.FAQ. Be advised that IP addresses whitelisted with PortKnocker (only!) go away whenever your server is rebooted or the IPtables firewall is restarted. For further information on the PortKnocker technology and available clients for iOS and Android devices, review the Nerd Vittles tutorial.

HINT: The reason that storing your PortKnocker codes in a safe place is essential is because it may be your only available way to gain access to your server if your IP address changes. You obviously can’t use the command-line tools to whitelist a new IP address if you cannot gain access to your server at the new IP address.

We always recommend you also add an extra layer of protection by running your server behind a hardware-based firewall with no Internet port exposure, but that’s your call. If you use a hardware-based firewall, be sure to map the three PortKnocker ports to the internal IP address of your server!

The NeoRouter VPN client also is included for rock-solid, secure connectivity for remote users. Read our previous tutorial for setup instructions.

As one would expect, the IPtables firewall is a complex piece of software. If you need assistance configuring it, visit the PIAF Forum for some friendly assistance.

Adding Incredible Fax 11 to Your Server

Once you’ve completed the Incredible PBX install, log out and log back in to load the latest automatic updates. Then reboot. Now you’re ready to continue your adventure by installing Incredible Fax 11 for Ubuntu. Special thanks to Josh North for all his hard work on this! The latest download includes the Incredible Fax 11 installer. So just run the script:

cd /root
./incrediblefax11_ubuntu14.sh

Accept all of the defaults during the installation process. IMPORTANT: Once you complete the install, reboot your server. After rebooting, log into the GUI and choose Module Admin and enable the AvantFax module. When you log out of the GUI, there now will be an option for AvantFax on the GUI’s main login screen. Choose it and enter admin:password to login and change your default password. You also can set your AvantFax admin password by logging into the Linux CLI and… /root/avantfax-pw-change.

Incredible Backup and Restore

We’re pleased to introduce our latest backup and restore utilities for Incredible PBX. Running /root/incrediblebackup will create a backup image of your server in /tmp. This backup image then can be copied to any other medium desired for storage. To restore it to another Incredible PBX 11 server, simply copy the image to a server running Asterisk 11 and the Incredible PBX 11-12 GUI. Then run /root/incrediblerestore. Doesn’t get much simpler than that.

Incredible PBX Automatic Update Utility

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

Mastering the Incredible PBX Applications

Your next stop should be a quick read of the Application User’s Guide for Incredible PBX. Even though the target audience was Raspberry Pi users, the feature set is identical, and this guide will tell you everything you need to know about the dozens of applications for Asterisk that have been installed on your new server.

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

Originally published: Monday, June 1, 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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Some Recent Nerd Vittles Articles of Interest…

  1. With some providers including ones linked in this article, Nerd Vittles receives referral fees which assist in keeping the Nerd Vittles lights burning brightly. []

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 Gotcha-Free PBX: GVsip Gateway Service 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 today the first of two SIP gateway offerings has arrived. With this new service, you simply create a standard SIP trunk on your Asterisk server of choice, associate your Google Voice account with the GVsip gateway service, and PRESTO! You get secure OAUTH 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 without compromising your credentials for a one-time setup fee of $20 (yes, the price quadrupled!).

NEWS FLASHES: The second SIP Gateway for Google Voice has just been released by Simon Telephonics. Our review is available here. GVsip now includes Voice Dialing! Dial 1 or * from your GVsip trunk. At the tone, say: "Dial 18005551212"

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 to the GVsip gateway 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.

Do us the favor of using our signup link for the new GVsip gateway service so that Nerd Vittles gets a piece of the action to keep the lights on. If you’re one that never trusts too-good-to-be-true offers, then take advantage of the free trial without ever pulling out your credit card. So here’s how to get started.

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 GVsip gateway. Be sure that you’ve enabled Google Chat in your Google Voice setup.

2. Using a separate tab of your browser, click on the Get Access / Login with Google button on the GVsip site.

3. Go through the steps to associate your Google Voice account with the GVsip gateway and obtain 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 actual GVsip credentials (replace ACCTNO and ACCTPW) and Google Voice number (replace 8005551212):

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

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

Once your trunk is up and running, skip sections 5 and 6 below and jump to Step #7 to complete the install.

5. Create an Inbound Route for your incoming calls using the 10-digit number you entered in the previous step.

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.

7. Finally, go back to the GVsip site and login again if your original login expired. Then associate your registered GVsip trunk with your Google Voice account after accepting the Terms of Service agreement.

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=GVsip entry to something like TRUNK=GVsip2. 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: Friday, April 3, 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.


NEWS FLASH: The Grandstream HT701 Handytone 701 ATA Analog Telephone Adapter with Lifetime Subscription to GVsip has just been released. For those with standard POTS phones, this ATA at $29.99 is a terrific Google Voice solution. Using our Amazon referral link helps keep the Nerd Vittles lights burning brightly.



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


 
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