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Monthly Archives: April 2015

The Most Versatile VoIP Provider: FREE PORTING

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:


https://youtu.be/Bp_7m64k_ko

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.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

Gotcha-Free PBX: GIT-R-Done with Incredible PBX for Asterisk-GUI (CentOS)


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For the die-hard developers out there, we are pleased to introduce a new version of Incredible PBX™ for Asterisk-GUI that uses GIT repos to build both Asterisk® and Asterisk-GUI with the same feature set of applications as our previous releases. You still get a Gotcha-Free PBX with pure and honest open source GPL code. No patent, trademark, or copyright minefields to trip you up. But this time around you’ll have an Asterisk platform that can be updated in seconds by running a simple upgrade script: upgrade-asterisk-to-current. Special thanks to Matt Jordan & Co. for the new GIT implementation. And our extra special thanks to Denver sports cartoonist, Drew Litton, for letting us share his GIT-R-DONE creation as well.

This time around you’ll need a 64-bit CentOS 6.5/6.6 base platform. When you complete the 30-minute install procedure, you’ll have the very latest version of Asterisk 11 and Asterisk-GUI. Both are compiled from source on your hardware platform to maximize performance. The end result is the VoIP Trifecta… better, cheaper, and faster.

Since the early Windows® days, we haven’t been big fans of GUI-only interfaces. Let’s face it. Some things can be configured more efficiently with less chance for error using other tools. Incredible PBX takes advantage of this hybrid technology by offering the best of all worlds. Administrators can use a GUI where it makes sense and use a text editor or simple web form where it doesn’t. There’s no MySQL middleware to obfuscate your Asterisk settings. So you can configure 8 VoIP trunks from 8 great providers in under 5 minutes. And there’s so much more…

Target Audience: Home or SOHO/SBO in need of a turnkey, Gotcha-Free PBX Development Platform

Default Configuration: Asterisk 11 with enhanced Asterisk-GUI, Kennonsoft GUI, and NANPA dialplan

Platform: 64-bit CentOS 6.5/6.6 running on Dedicated Server, Cloud-Based Server, or Virtual Machine

Minimum Memory: 512MB

Recommended Disk: 20GB+

Default Trunks: Google Voice, CallCentric, DIDlogic, Future-Nine, IPcomms, Les.net, Vitelity, VoIP.ms1

Feature Set: Fax, SMS messaging, 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, OpenCNAM CallerID lookups, DISA, Call Forwarding, CSV CDRs

Administrator Utilities: Incredible Backup/Restore, Automatic Updater, Asterisk Upgrader, phpMyAdmin, Timezone Config, Plug-and-Play Trunk Configurator, WebMin, External IP Setup, Firewall WhiteList Tools

Getting Started with Incredible PBX for Asterisk-GUI (GIT Edition)

Here’s a quick overview of the installation and setup process for Incredible PBX for Asterisk-GUI:

  1. Choose a Hardware Platform – Dedicated PC, Cloud, or Virtual Machine
  2. Install Linux – 64-bit CentOS 6.5 or Scientific Linux Minimal ISO
  3. Download and Install Incredible PBX for Asterisk-GUI
  4. Install Incredible Fax for Asterisk-GUI (optional)
  5. Set Up Passwords for Incredible PBX for Asterisk-GUI
  6. Configure Trunks with Incredible PBX for Asterisk-GUI
  7. Connect a Softphone to Incredible PBX for Asterisk-GUI

1. Choose a Platform for Incredible PBX for Asterisk-GUI

Incredible PBX for Asterisk-GUI works equally well on dedicated hardware 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 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 Implementations, this time around we recommend Digital Ocean because the GIT edition is designed to be a development platform with bleeding edge Asterisk 11 code.

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.

2. Install a Linux Flavor for Incredible PBX for Asterisk-GUI

To be clear, we plan to support many Linux flavors other than RedHat. But Rome wasn’t built in a day so hang in there. We’re flippin’ burgers as fast as we can. For today, you’ll need a 64-bit version of CentOS or Scientific Linux 6.5/6.6. On some platforms, you install 6.5. After the initial update and upgrade steps, you’ll end up with 6.6. There are many flavors of CentOS and Scientific Linux. For Incredible PBX, a minimal install is all you need.

With dedicated hardware, begin by downloading the 64-bit CentOS 6.6 minimal ISO. Boot your server with the ISO, and begin the install. Here are the simplest installation steps:

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

With most cloud-based providers, you simply choose the CentOS 6.5 platform in creating your initial image. 512MB of RAM is plenty so long as you have a swap file. Within a minute or two, you’re ready to boot up the server.

For VirtualBox, download the Scientific Linux 6.6 minimal install .ova image from SourceForge. Then double-click on the image to load it into VirtualBox. Enable Audio and configure Network with Bridge Adapter in Settings. Then start the virtual machine. Default password for root is password.

With VirtualBox, you can skip this step. For everyone else, log into your server as root and issue the following commands to put the basic pieces in place and to reconfigure your Ethernet port as eth0. On some platforms, some of the commands may generate errors. Don’t worry about it! Just make a note of your IP address so you can log back in with SSH from a desktop computer to begin the Incredible PBX install.

For CentOS/Scientific Linux 6.5 minimal install:

setenforce 0
yum -y upgrade
yum -y install net-tools nano wget
ifconfig
sed -i 's|quiet|quiet net.ifnames=0 biosdevdame=0|' /etc/default/grub
grub2-mkconfig -o /boot/grub2/grub.cfg
wget http://incrediblepbx.com/update-kernel-devel
chmod +x update-kernel-devel
./update-kernel-devel
reboot

For CentOS/Scientific Linux 6.6 minimal install:

setenforce 0
yum -y upgrade
yum -y install net-tools nano wget
ifconfig
reboot

3. Install GIT-R-Done Edition of Incredible PBX for Asterisk-GUI

cd /root
yum -y install wget
wget http://incrediblepbx.com/incrediblepbx11gui-git.tar.gz
tar zxvf incrediblepbx11gui-git.tar.gz
#./create-swapfile-DO  #add this step for Digital Ocean droplets
rm -f incrediblepbx11gui-git.tar.gz
sed -i 's|pbxinaflash.com|incrediblepbx.com|' IncrediblePBX11-GUI-git.sh
./IncrediblePBX11-GUI-git.sh
./IncrediblePBX11-GUI-git.sh

4. Install Incredible Fax for Asterisk-GUI (optional)

Administrators have been trying to stomp out faxing for at least two decades. Here’s a hint. It ain’t gonna happen. So go with the flow and add Gotcha-Free Faxing to your server. It’ll be there when you need it. And sooner or later, you’ll need it. This install script is simple enough for any monkey to complete. Run the script and enter the email address for delivery of your faxes. Then, if you’re in the U.S. or Canada, press the Enter key to accept every default entry during the HylaFax and AvantFax installation steps. For other countries, read the prompts and answer accordingly. When the installation finishes, reboot your server to bring faxing on line. Be sure to change your AvantFax admin password. By default, it is password. You can use the script included in the /root folder: avantfax-pw-change. REMINDER: Don’t forget to reboot your server!

cd /root
./incrediblefax11-GUI.sh
./avantfax-pw-change
reboot

Troubleshooting: If your IAXmodems don’t display with a green IDLE notation in the AvantFax GUI, you may need to restart them once more. After a second reboot, all should be well. The restart command is /root/iaxmodem-restart.

5. Initial Configuration of Incredible PBX for Asterisk-GUI

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, Asterisk-GUI has its own security methodology using Asterisk’s manager.conf. 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 10 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 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
Make a copy of your Knock codes: cat knock.FAQ
Decipher IP address and other info about your server: status

Second, log into your server as admin using a web browser pointed to your server’s IP address:

Click USERS tab in Incredible PBX GUI
Click Asterisk-GUI Administration
Log in as user: admin with password: password
Immediately change your admin password and login again

Log in to Asterisk-GUI again with your new password. Expand the options available in the GUI:

Options -> Advanced Options -> Show Advanced Options

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. 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.

6. Configure Trunks with Incredible PBX for Asterisk-GUI

Now for the fun part. If this is your first VoIP adventure, be advised that this ain’t your grandma’s phone system. You need not and should not put all your eggs in one basket when it comes to telephone providers. In order to connect to Plain Old Telephones, you still need at least one provider. But there is nothing wrong with having several. And a provider that handles an outbound call (termination) need not be the same one that handles an incoming call (origination) and provides your phone number (DID). We cannot recommend Vitelity highly enough, and it’s not just because they have financially supported our projects for almost a decade. They’re as good as VoIP providers get, and we use lots of them. If you’re lucky enough to live in the U.S., you’d be crazy not to set up a Google Voice account. It’s free as are all phone calls to anywhere in the U.S. and Canada. The remaining preconfigured providers included in Incredible PBX for Asterisk-GUI are equally good, and we’ve used and continue to use almost all of them. So pick a few and sign up. You only pay for the calls you make with each provider so you have little to lose by choosing several. The PIAF Forum includes dozens of recommendations on VoIP providers if you want additional information.

With the preconfigured trunks in Incredible PBX for Asterisk-GUI, all you need are your credentials for each provider and the FQDN of their server. Log into Asterisk-GUI Administration as admin using a browser. From the System Status screen, click Incredible PBX Apps. Click on each provider you have chosen and fill in the blanks with your credentials. When you’ve saved all of your settings, log into your server as root via SSH and type: service asterisk restart or asterisk-restart. You can also issue the command in the Asterisk-GUI by choosing the Asterisk CLI tab2 in the left column. Doesn’t get any simpler!

Update: It should be noted that Incredible PBX for Asterisk-GUI also supports Anveo Direct trunks; however, they are configured differently because of the way Anveo handles the calls. You’ll need the PIN provided by Anveo to set up your trunk, and Anveo supports CallerID spoofing so you can enter any CallerID number for the trunk that you are authorized to use. You’ll find the Anveo Direct setup link in the Incredible PBX Apps tab. To route an outgoing call through Anveo trunk, dial 2 + any desired 10-digit number.

Here is the complete list of dialing prefixes and the trunks to which they are associated:

  • 1 – Google Voice
  • 2 – Anveo Direct
  • 3 – Future Nine
  • 4 – CallCentric
  • 5 – DIDlogic
  • 6 – IPcomms
  • 7 – Les.net
  • 8 – Vitelity
  • 9 – VoIP.ms

For free iNUM calling worldwide, the following dialing prefixes are supported in conjunction with the last seven digits of any destination iNUM DID. Free iNUM DIDs for your own PBX are available from both of these providers as well.

  • 0XXXXXXX – CallCentric
  • 90XXXXXXX – VoIP.ms

Finally, in addition to the native Asterisk motif implementation of Google Voice (covered below) which uses insecure authentication with Google Voice, we also support the new Simonics SIP gateway to Google Voice using OAUTH authentication. Just click this link for the installation script and tutorial.

7. Configure a Softphone with Incredible PBX for Asterisk-GUI

We’re in the home stretch now. You can connect virtually any kind of telephone to your new Gotcha-Free 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 telephony.

We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 6002 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 6002 password. Choose Users -> 6002 and write down your SIP/IAX Password. You can also find it in /root/passwords.FAQ. Fill in the blanks using the IP address of your server, 6002 for your account name, and whatever password is assigned to the extension. Click OK to save your entries.

Once you are registered to extension 6002, 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:

7001 - IVR Demo
123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
*61 - Time of Day
TODAY - Today in History

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.

Configuring Google Voice

If you want to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX. 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!

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

Once you’ve created your Gmail and Google Voice accounts, go to Google Voice Settings and 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’ve adjusted 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 configure your Google Voice account in Incredible PBX. You can do it from within Asterisk-GUI by choosing Google Voice within the Incredible PBX Apps tab. Once you entered your credentials, don’t forget to restart Asterisk, or Google Voice calls will fail. If you still have trouble placing or receiving calls, try these tips.

OK, Smarty Pants: Show Me the Beef!

We know what some of you are thinking. "What does a fast food worker really know about VoIP and Gotcha-Free PBXs?? Before I waste a bunch of time on this, show me the beef!" Fair enough. Sit by your phone and click the Call Me icon below. Type in a fake name and your real phone number. Click the Connect button, answer your phone when it rings, and press 1. You’ll be connected to the Incredible PBX IVR for Asterisk-GUI. Pick an option from the menu of choices and take the Incredible PBX apps for a spin on our dime… actually it’s Google’s dime. Everything you see and hear is part of what you get with Incredible PBX for Asterisk-GUI including the ability to set up your own click-to-dial web interface exactly like this one. The demo just happens to be running on our Mac desktop instead of yours. So… what are you waiting for? Click away and try Incredible PBX for yourself. And, by the way, nobody besides the NSA and Google will be monitoring your call. 😉



Nerd Vittles Demo IVR Options
1 – Call by Name (say “Delta Airlines” or “American Airlines” to try it out)
2 – MeetMe Conference (password is 1234)
3 – Wolfram Alpha (say “What planes are overhead?”)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (say the city and state, province, or country)
7 – Today in History
8 – Speak to a Real Person (or maybe just voicemail if we’re out)

Homework Assignment: Mastering the Asterisk-GUI

We’ll have more to say about the Incredible PBX applications next week. In the meantime, you have some homework. You need to learn all about Asterisk-GUI and how to make the best use of its powerful feature set. Here’s one word of warning. We mentioned that Incredible PBX was a hybrid system that combines some customized settings with the standard Asterisk-GUI interface. Before modifying existing settings for the default trunks, extensions, and default routes, take a look at the credentials* files in /etc/asterisk. If you modify any of these trunk entries or the Outgoing or Incoming Call Rules in Asterisk-GUI, you may break the Incredible PBX setup. So steer clear of that minefield until you know what you’re doing. Adding new extensions and additional trunks is perfectly fine and will not break anything.

Rather than reinvent the wheel, we’ll point you to some excellent tutorials that already have been written. Start with Chapter 3 of Digium’s Asterisk Appliance™ Administrator Manual. Next, review Chapter 11 of The Asterisk Book (Second Edition). Finally, take a look at a couple of the tutorials that have been written by other companies that incorporated Asterisk-GUI into their hardware products, e.g. Yeastar’s MyPBX SOHO User Manual and Grandstream’s UCM6100 User Manual. Then check back with us next week for Chapter 2.

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.

We also are quickly building a collection of tutorials tailored specifically for Incredible PBX for Asterisk-GUI:

Enjoy your new Gotcha-Free PBX!

Now Available: The Gotcha-Free Incredible PBX Application User’s Guide

Originally published: Monday, April 20, 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.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

  1. Vitelity and Google provide financial support to Nerd Vittles and the Incredible PBX project. []
  2. If, for some reason, the Asterisk CLI tab does not appear on your server, click Options -> Advanced Options -> Show Advanced Options. []

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.


 

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



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  1. In addition to substantial technical assistance, Simon Telephonics is also a financial contributor to the Nerd Vittles project. []