Posts tagged: google voice

60 Seconds to Real Independence: Incredible PBX GUI Comes to VirtualBox

As we continue the march toward a truly free, RealGPL, open source VoIP platform for Asterisk®, we couldn’t think of a better time of the year for this announcement. Today we’re pleased to introduce our first virtual machine platform with an Incredible PBX™ GUI image that you can install in less than 60 seconds on virtually any desktop computer in the world. When the install is finished you’ll have the latest Asterisk 11 running atop Scientific Linux™ 6.6 with version 12 of the new Incredible PBX GUI. You’ll also have the very first Asterisk aggregation with native support for OAUTH authentication and secure communications using Google Voice. And it’s all FREE. No Gotchas!

Think of Incredible PBX as the glue stick that assembles all the necessary VoIP components into a state of the art Linux platform and holds them together seamlessly. As with all Incredible PBX builds, you also get the full complement of goodies including dozens of text-to-speech apps, voice dialing, SMS messaging, free fax support, reminders and wakeup calls, and SECURITY! The difference with the VirtualBox® platform is you get a turnkey install of everything on any desktop computer in less than one minute! That includes Windows PCs, Macs, Linux desktops, and even Solaris machines.

Is VirtualBox merely a sandbox for experimentation? Absolutely not. With any of the beefier desktop computers today, running Incredible PBX as a 24/7 VirtualBox image is every bit as feature rich with stellar performance, and it’s equivalent to using dedicated hardware. And there are some added advantages. Obviously, deploying a turnkey VoIP platform in under a minute is a major plus. But, unlike using a dedicated Linux platform, you also get the ability to take snapshots of your system and do full backups in minutes instead of the hours required to bring down dedicated hardware, load a different backup application using a different operating system, perform a backup, and then reboot your VoIP server. And your backups won’t just run on the one server on which the backup was performed. You can restore the backup to any other computer that can run VirtualBox. For any of you that came from a network management background, you know what a big deal that really is. And there’s one more bonus. With Incredible Backup and Restore, you can move to dedicated hardware running the same operating system with Asterisk 11 and the same version of the Incredible PBX GUI in minutes.

Need to deploy VoIP servers at dozens of sites around the globe? Not a problem with VirtualBox. Just send a preconfigured VirtualBox image to each site and install VirtualBox on a local desktop computer. In 60 seconds, you’ll have a functional VoIP server including interconnectivity to all of your other VoIP servers with a virtual private network already in place to provide secure VoIP connectivity between all of your sites.

Are there security compromises using the VirtualBox platform? Not at all. Incredible PBX comes preconfigured with the Linux IPtables firewall that is locked down to a whitelist of local area networks, preferred providers, and your own IP addresses. You can expand the whitelist using the add-ip and add-fqdn scripts or use PortKnocker and Travelin’ Man 4 tools to let remote users gain instant access.

So What’s All the GPL Fuss About? It’s about FREEDOM, the freedom to use or not use the GPL modules you wish to use without enduring false alerts that your system has been compromised and without being blocked from removing components that produce revenue for Sangoma®… as the GPL requires. It’s about FREEDOM to redistribute or resell the product AS IS… as the GPL requires. It’s about FREEDOM to examine and modify ALL of the source code using ALL of the tools and components necessary, not just ones Sangoma has chosen to provide… as the GPL requires. It’s about FREEDOM to add GUI components to your server with No Gotchas whether or not the individual modules were produced by Sangoma… as the GPL requires.


If you support the GPL and use open source projects, then you owe it to yourself and to the GPL community to get up to speed and get involved! Can’t we all just get along? You bet… when everyone does what they’ve agreed to do. Spend an hour or two of your Independence Day reading some of the Nerd Vittles commentary on FreePBX® and the GPL.

BUY 3 STAMPS and let Sangoma and Digium hear from you. Don’t be shy. It’s about your FREEDOM.

William J. Wignall, President and CEO
Sangoma Technologies
100 Renfrew Drive, Suite 100
Markham ON L3R 9R6 CANADA

Danny Windham, CEO
Digium, Inc.
445 Jan Davis Drive Northwest
Huntsville, AL 35806 USA

Mark Spencer, Founder and CTO
Digium, Inc.
445 Jan Davis Drive Northwest
Huntsville, AL 35806 USA

Getting Started. For today, we’ll provide a refresher course on loading VirtualBox and the Incredible PBX virtual image. Then we want to spend a little time explaining the secret sauce that goes into building these images so that you can do it yourself either to migrate to a different network or to deploy at multiple sites. It’s called open source for a reason! When we’re finished, you’ll know everything we’ve learned about deploying VirtualBox machines and, unlike Grandma and some GUI platforms, we won’t leave an important ingredient out of the recipe just to be sure you never forget how good Grandma’s cookies really were. So let’s get started.

Installing Oracle VM VirtualBox

Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.

Downloading the Incredible PBX GUI Virtual Machine

A word of warning on the front end. Today’s Incredible PBX image featuring Asterisk 11 for VirtualBox is huge! The Scientific Linux 6.6 image with version 12 of Incredible PBX GUI is nearly 3GB. Be patient. You only have to download it once. Just click on the 11-12.3 .OVA image in this SourceForge link and start the download to your desktop. Then go have a nice lunch.

Importing & Configuring Incredible PBX Virtual Machines in VirtualBox

You only perform the import step one time. Once imported into VirtualBox, Incredible PBX is ready to use. There’s no further installation required, just like an OpenVZ template… only better. Double-click on the .ova file you downloaded to begin the procedure and load it into VirtualBox. When prompted, be sure to check the Reinitialize the Mac address of all network cards box and then click the Import button. Once the import is finished, you’ll see a new Incredible PBX virtual machine in your VM List on the VirtualBox Manager Window. We need to make a couple of one-time adjustments to the Incredible PBX VM configuration to account for differences in sound and network cards on different host machines.

Click on the Incredible PBX Virtual Machine in the VM List. Then click Settings -> Audio and check the Enable Audio option and choose your sound card. Save your setup by clicking the OK button. Next click Settings -> Network. For Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. Finally, click Settings -> System, uncheck Hardware clock in UTC time, and click OK. That’s all the configuration that is necessary for your Incredible PBX Virtual Machine. The rest is automagic.

Running Incredible PBX Virtual Machines in VirtualBox

Once you’ve imported and configured the Incredible PBX Virtual Machine, you’re ready to go. Highlight IncrediblePBX Virtual Machine in the VM List on the VirtualBox Manager Window and click the Start button. The boot procedure with your chosen operating system will begin just as if you had installed Incredible PBX on a standalone machine. You’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VirtualBox window. Always gracefully halt Incredible PBX just as you would on a dedicated computer.

Here’s what you need to know. To work in the Incredible PBX Virtual Machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. To access Incredible PBX GUI with a browser, point to the IP address of your virtual machine. Then, in the Administrator window, click on Incredible GUI Administration. Login as admin with the admin password you set below. For the security of your server, we recommend that you log in to the Linux CLI at least once a week so that Incredible PBX updates get applied to your server regularly. This is critically important if you care about your phone bill.

When logging in for the first time, Incredible PBX will go through some setup steps and then reboot. Login again to complete the setup. status will always provide a snapshot of your system. To shut down Incredible PBX gracefully, click in the VM window with your mouse, log in as root, and type: halt. Be sure to complete the following setup steps from the Linux CLI:

  • Change your root password: passwd
  • Set your Incredible GUI admin password: /root/admin-pw-change
  • Set the admin password for web apps: htpasswd /etc/pbx/wwwpasswd admin
  • Set your correct time zone: /root/timezone-setup
  • Add WhiteList entries to firewall if needed: /root/add-ip or /root/add-fqdn
  • Store PortKnocker credentials in a safe place: cat /root/knock.FAQ
  • Enable Incredible Fax support if desired: /root/incrediblefax11.sh
  • Login to your NeoRouter VPN server if desired: /root/neorouter-login

Upgrading Modules with Module Admin in the GUI. The GUI includes a Module Administration component in the Admin tab which will let you check online for new modules and upgrade to newer releases. Once you have added or updated any modules, you will get some nasty error messages in the System Status display because we allow installation of all GPL-compatible modules, not just those of Sangoma. It’s one of the proprietary gotchas that we have been writing about. Simply click on the X option in the upper right corner of each window to remove the warnings. Log out of the GUI. Then login to your Linux CLI as root and issue the following command to permanently clear the error messages: gui-fix. Now you can log back in and the warning messages will be gone… until you add or update modules again. Sangoma calls it a feature. :roll:

Command Line Management of Incredible PBX with VirtualBox

One of the real beauties of VirtualBox is you don’t have to use the VirtualBox GUI at all. The entire process can be driven from the command line. Other than on a Mac, here is the procedure to import, configure, and run Incredible PBX:
 
VBoxManage import IncrediblePBX-11-12.3-SL66.ova
VBoxManage modifyvm "IncrediblePBX-11-12.3-SL66" --nic1 nat
VBoxManage modifyvm "IncrediblePBX-11-12.3-SL66" --acpi on --nic1 bridged
VBoxHeadless --startvm "IncrediblePBX-11-12.3-SL66" &
# Wait 1 minute for Incredible PBX to load. Then decipher IP address like this:
VBoxManage guestproperty get "IncrediblePBX-11-12.3-SL66" /VirtualBox/GuestInfo/Net/0/V4/IP
# Now you can use SSH to login to Incredible PBX at the displayed IP address
# Shutdown the Incredible PBX Virtual Machine with the following command:
VBoxManage controlvm "IncrediblePBX-11-12.3-SL66" acpipowerbutton

On a Mac, everything works the same way except for deciphering the IP address. Download our findip script for that. Be sure to plug in the correct name of your virtual machine: ./findip IncrediblePBX-11-12.3-SL66

Deploying Google Voice Secure Communications with Incredible PBX

As with all prior releases of Incredible PBX, free calling in the U.S. and Canada with Google Voice is an integral component of this GPL platform. You still add Google Voice trunks using the GUI in exactly the same way: Connectivity -> Google Voice (Motif). What has changed under the covers with this release is what happens behind the scenes. Google has warned (for years) that they plan to phase out plain text passwords using your actual Google Voice credentials. This is for your protection! Unfortunately, until today, the only way to take advantage of the new OAUTH authentication method with Asterisk was to use one of the external SIP gateways to Google Voice. Now you no longer have to. The new 11-12.3 release of Incredible PBX adds native OAUTH authentication support to Asterisk and the Incredible PBX GUI. When prompted for the password in setting up your Google Voice accounts in the GUI, now you’ll enter your OAUTH token instead of your plain text password. It’s that easy. Obviously, you first need to obtain a free OAUTH token for each of your Google Voice accounts that you wish to activate. This tutorial on the PIAF Forum will walk you through the simple, one-time procedure.

IMPORTANT: Once you have added one or more Google Voice trunks in the GUI, you must restart Asterisk to activate the trunks: amportal restart

We want to take a moment and express our heartfelt thanks to Ryan Tilton of GVsip.com for setting up and maintaining the free platform to support OAUTH tokens for Google Voice. And a special shoutout to Martin Dindos (a.k.a. @dziny on the PIAF Forum) for his truly Herculean efforts in getting this to work properly with Asterisk 11, no small feat. This is yet another amazing testament to how the open source community should really function. Thank you!

Preparing Incredible PBX Virtual Machine for Backups & Migration

To us, the most compelling feature of the virtual machine platform is the ease with which you can make a perfect backup of your server in minutes! From that backup, you can restore a working platform in the same 60 seconds it took to build today’s platform on your desktop. One of the drawbacks as the Linux operating systems have become more turnkey is the shortcut that was implemented on both the RedHat and Debian/Ubuntu platforms to store your network setup so that the server reboots more quickly. While that’s fine for rebooting on the same server, it’s a real problem if you attempt to move your setup to different hardware or a new network because your network configuration will not load properly on the new platform. That means no IP address! Here’s the easy way to assure that things will actually work after the move. It assumes you will have a DHCP server at the new location just as you did at your existing site.

The Easy Way. If you have console access after the VM image is restored on the new platform (which means you don’t need a network IP address for the server in order to log in as root), then the easy way to prepare any of the Incredible PBX machines for relocation is to issue the following commands before you halt the system and make a VirtualBox backup:

touch /etc/update_hostconfig
touch /etc/update_serverconfig
rm -f /etc/ssh/ssh_host*
rpm -e openssh-server openssh-xinetd
yum -y install openssh-server openssh-xinetd
rm /etc/ssh/*.rpmsave

Once you have halted the server, edit both the sound card and network card settings and disable both of them in VirtualBox Manager. Then choose File -> Export Appliance from the VirtualBox title bar and create an .ova backup image on your desktop. You now have an image that is similar to the Incredible PBX image that you originally downloaded, except it has all of your data and settings. All you have to do is repeat the install drill above at the new location using the .ova image you created and log in with whatever your current root password happens to be. You’ll get a two-pass automatic setup just as you did when you began today’s adventure.

The only drawback to this procedure is the fact that the extension 701 and default DISA passwords as well as your firewall configuration will be initialized when you first boot from your .ova image at the other location. Aside from that, you’ll have a clean platform with new SSH and DUNDI credentials as well as mostly sanitized log files.

What’s Next. Now that you have a functioning server, it’s time to learn all about the Incredible PBX applications that are ready for use. Jump over to the latest Nerd Vittles application tutorial for a quick look at what’s available. Even though it was written for the Asterisk-GUI, everything will work exactly the same way. That’s the beauty of the Incredible PBX platform. Enjoy!

Originally published: Monday, June 29, 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. Our forum is extremely friendly and is supported by literally hundreds of Asterisk gurus.



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. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. []

Decisions, Decisions: Choosing the SOHO Asterisk Platform That’s Best For You

Each year we like to revisit the topic of choosing the best Asterisk® platform for deployment in the home and small business environment. No solution is obviously right for everybody. But we think it’s important to sketch out the relevant factors that need careful evaluation before you begin the installation process.

Our focus today is open source, GPL platforms with Asterisk for home or SOHO deployments. That excludes a broad swath of equally capable commercial or proprietary alternatives including ThirdLane, Switchvox, and FreePBX® Distro as well as many unified communications solutions that do not rely upon the Asterisk telephony engine including FreeSWITCH, ShoreTel, Cisco, 3CX, and many others. If your requirements exceed telephony support for more than a few dozen employees, our recommendation is to hire a consultant that can assist you in that decision-making process.

When It Comes to Hardware, Size Matters!

Even in the telephony world, it’s true. Size Matters! Choosing an Asterisk platform for your home and choosing a telephony platform for a call center are very different beasts. Our traditional recommendation for home and SOHO deployments was to go with dedicated hardware with an appropriately sized Atom processor, RAM, and hard drive. In the words of Bob Dylan, “The Times They Are A Changin’.” With the nosedive in Cloud processing costs and the emergence of powerful desktop virtual machine platforms, that may no longer be the smartest solution. First, it puts you in the hardware business which means you’ll have to deal with hardware failures and backups and redundancy. Second, depending upon where you live, it may not be cost-effective to maintain your own server. Electricity and Internet connectivity cost real money above and beyond hardware costs.

For home or SOHO deployments, it also depends upon what other computers already are in use around your house or office. For example, if you have a $2,000 iMac with a $100 backup drive running Carbon Copy Cloner each night, then you’ve already got a fully redundant server platform in place. You really don’t need a dedicated server for telephony to support a handful of telephones. VirtualBox® running any of the Incredible PBX™ solutions is free, and it’s fully capable of meeting your telephony requirements with no additional hardware investment.1 If your iMac’s main drive crashes, you can reboot from the attached USB backup drive with a single keystroke and never miss a beat. For those dead set on running dedicated hardware for your home or SOHO telephone system, there’s really no reason to spend more than $35 for a Raspberry Pi® 2. With its new quadcore processor and gig of RAM, it can meet or exceed any requirements you may have. Buy a second microSD card for redundancy and call it day as far as hardware is concerned.

If you’d prefer to separate your telephone system from your house or small office, a Cloud-based setup may be a better fit. Our Platinum sponsor, RentPBX,2 offers a worldwide collection of servers and will host your Asterisk-based PBX for $15 a month (Coupon Code: NOGOTCHAS) on a platform that rarely, if ever, goes down. If you like to tinker but also prefer a Cloud solution, consider Digital Ocean ($5 a month for a virtual machine) or Wable ($8 a month for up to 5 VMs).

NEWS FLASH: Effective today, RentPBX now offers all of the new Incredible PBX builds with the Incredible PBX GUI. Tutorials available here: CentOS platform or Ubuntu platform. Use the NOGOTCHAS coupon code for $15/mo. pricing.

That’s our latest take on SOHO hardware. If you have additional questions or concerns, come join the PIAF Forum and take advantage of our hundreds of gurus who will give you all of the free advice you could ever want.

I’ve Got My Hardware Platform. Now What?

The next step is choosing an Asterisk telephony platform. That used to be easy. There was Plain Ol’ Asterisk if you were a guru or there was Asterisk@Home if you wanted a GUI to guide you through the telephony maze. Now it’s more complicated. There are a number of different Linux platforms. There are a number of different Asterisk versions. And there are a number of different GUIs that support Asterisk. So let’s work our way down the list starting with the Linux platform.

Choosing the Linux Platform That’s Best for Asterisk

The gold standard for Asterisk servers has always been CentOS, a GPL clone of RedHat Enterprise Linux. It, too, is now owned by Red Hat. The old adage was that nobody ever got fired for recommending IBM. In the Asterisk community, that remains true with CentOS. Unfortunately, CentOS now comes in several flavors. There’s CentOS 6.5 which morphs into 6.6 once the latest updates are applied. Or there’s CentOS 7 which is a very different beast. For Asterisk deployments, you can’t go wrong with CentOS 6.5. It works well on the latest dedicated hardware and is supported on all virtual machine platforms.

As with choosing a language, you now have a choice of Linux platforms. There’s RedHat/CentOS, or Debian, or Ubuntu, or even Raspbian for the Raspberry Pi hardware. Unfortunately, the RedHat-CentOS and Debian-Ubuntu-Raspbian platforms have completely different languages, much like French and Spanish. The Linux packages that are included in the platforms also have different names. If you’re a Linux aficionado and you already have a favorite, stick with what you love. If you’re planning to deploy a Raspberry Pi 2, stick with Raspbian. For everyone else, CentOS 6.5 is your best bet for now.

Choosing the Asterisk Platform That’s Right for You

Believe it or not, there are many organizations still running their telephone systems using Asterisk 1.4 or 1.8 even though Digium support for those platforms ended years ago. In the commercial world, it is not uncommon to see telephone systems that are more than a decade old. With Asterisk, things are quite different. There’s a new version every year. Fortunately, Digium has adopted a new support philosophy and every other release now is anointed with the LTS (Long Term Support) moniker. An LTS release gets four years of bug fixes and five years of security updates as opposed to the other releases that come with one year of bug fixes and two years of security updates. It’s still not 10 years, but it’s certainly better than wrestling with Asterisk updates annually.

We think there remains a need to reconsider these timetables. New updates have become so complex that the releases typically are almost two years into their life cycle before there is anyone that treats the releases as anything more than experimental. This was especially true of Asterisk 12 which was a terrific new product that provided dramatic improvements particularly in the SIP area. Unfortunately, it will reach end-of-life status before the end of this year and before most folks have even had an opportunity to use it. Now we’re on to Asterisk 13 which almost no one has deployed, and it will be a year old this fall.

Choosing an Asterisk release has been further complicated by Sangoma’s shenanigans regarding FreePBX® 12, the only GUI platform that currently supports both Asterisk 12 and 13. If you want to deploy a commercial FreePBX module not sold by Sangoma, you’re out of luck with FreePBX 12 despite the clear language of the GPL license. If you want to deploy any GPL open source module for FreePBX 12 other than those distributed by Sangoma, you’re bombarded with nasty security alerts suggesting that your server has been compromised. We won’t beat the dead horse. There are plenty of Nerd Vittles articles to fill in the details if you are interested in the background. Suffice it to say, it is having an impact on the decision many users and companies make concerning their Asterisk platform. If you want to avoid the CrippleWare, you need to stick with FreePBX 2.11 which means that Asterisk 11 is the last supported LTS version for this platform. We continue to be an optimist, believing that Sangoma will come to their senses and figure all of this out sooner or later. But for now, that’s a snapshot of the current landscape.

Choosing a GPL-Compliant GUI That Meets Your Needs

All of the GUIs for Asterisk have one primary purpose. They are code generators for the Asterisk telephony engine, nothing more. With each of them, you can turn off your web server after using the graphical user interface, and your phone system will continue to work as designed. Imagine our surprise to learn that an Asterisk GUI developer was actually threatened by lawyers of another provider of GPL GUI software for Asterisk because both GUIs used similar GPL-generated Asterisk code.

The bogus claim was that, while the GUI platform itself was GPL-licensed code, the actual dialplan code generated by the GUI was not GPL-licensed and hence was copyright-protected as a derivative work. In other words, you can use our GUI for free but not the code that it generates. Since the sole purpose of the GUI is to generate code, guess what your GPL license actually got you… absolutely nothing of value. Try finding that in the fine print or the GPL license much less in any published decision on copyright law. Under this interpretation, every time you click that Apply Config button, you’re downloading and using copyrighted dialplan code without a license. Just think. Lawyers get paid to spew out this bull with a straight face! Imagine getting a toaster for your birthday and then learning that you can use it for anything except making toast. Makes you want to go to law school, doesn’t it? Can you guess who the players are? Thought so.

Sangoma, are you listening? Think we’re still making all of this up? Care to see the original text of the lawyer’s letter? Understand why we’re mad? Will it take condemnation from the Free Software Foundation to get you to clean things up??

For the rest of the story…

That, my friends, is the type of players we’re dealing with in the Asterisk “community” and it’s all about money. Lucky for all of you and us, the threats were ignored, and we now have the Elastix MT GUI that respects its GPL license. We, of course, have released our own free Incredible PBX GUI for CentOS, Ubuntu, and Raspbian without the proprietary signature checking mechanism and trademark minefields. It also employs the same GPL-licensed modules as FreePBX including a publicly-accessible Cloud component that meets the source code disclosure requirements of the GPL. The choice is all yours!

Introducing the 3-Click Platform Decision Tree for Asterisk

Now that you have the background, we want to provide a simple Decision Tree tool that will guide you through choosing the Asterisk GPL aggregation that best meets your needs. After you’ve made your selections, the utility will point you to the tutorials that will walk you through downloading, installing, and using the platform of your choice. Our fully-documented Asterisk Aggregation Guide also is available. Enjoy!

Originally published: Monday, June 22, 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…

  1. We will introduce the all-new Incredible PBX GUI platform for VirtualBox next week on Nerd Vittles. If you’re in a hurry, the Pioneer’s Edition now is available with a tutorial to get you started on the PIAF Forum. []
  2. Some of our links refer users to service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from some of these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. However, when pricing is comparable or availability is favorable, we support these providers because they support us. []

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

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
./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.


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


Some Recent Nerd Vittles Articles of Interest…

  1. Vitelity 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.


 
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.


 
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…

SOHO Delight: Introducing the Ultimate Asterisk Appliance for Under $30

We continue our journey to identify cost-effective, Gotcha-Free Asterisk® solutions. And, yes, we eat our own dog food! So this week we turn our attention to a real sleeper. It’s an Asterisk appliance with an almost unbelievable price and an even more incredible feature set. With the PBX in a Flash™ and Incredible PBX™ projects, we meet hundreds of thousands of new VoIP enthusiasts each year. But let’s face it. Even software products as simple to use as ours present a formidable challenge to some folks that are new to networking and dealing with complex hardware setups. There’s also the corner grocery store and the mom-and-pop restaurants and the shoe repair store and the tire store and the neighborhood bike shop that shouldn’t have to spend hundreds of dollars each month for basic phone service. And then there are those with a cabin in the mountains or a weekend beach house that just want a plug-and-play communications device that’s available when you need it. So this week’s VoIP solution is dedicated to those on a budget that have no interest in spending months learning the intricacies of VoIP technology. These folks just want basic phone service that works at an affordable price. Bells and whistles are nice but not if they add complexity or cost. And, boy, do we have an incredible find to share with you today. What you’ll need in addition to this Asterisk appliance is electricity and a working Internet connection with a router/firewall. That’s it.

WARNING: We do not recommend EVER connecting the JS-200FX directly to the Internet because of potential security issues with this older version of Asterisk.

We purchased our first JS-200FX Asterisk Appliance from X100P.com for $89.95 with $15 for shipping from the Far East. But others tipped us off that refurbished units (that means they’ve actually been tested and they work) are regularly available for considerably less cost. We’ve added a direct link to the manufacturer for your convenience. Either way, the JS-200FX is a steal. In addition to a router and firewall, the appliance includes two FXS ports to connect plain old telephones, integrated WiFi to connect softphones and SIP devices wirelessly, and best of all turnkey Google Voice support for two lines to make free calls in the United States and Canada. Because the Asterisk-GUI is an integral part of the appliance, setup time is under 5 minutes. And we’ll show you how. As we love to say, if you can handle slice-and-bake cookies, you can do this. So here’s the drill:

  1. Sign up for Google Voice service (do it twice for double the fun!)
  2. Boot and login to JS200-FX after connecting network cable from ETH2 to a computer
  3. Configure Networking and Connect CAT5 from ETH1 to Internet router
  4. Configure Google Voice and Make a Call
  5. Configure Asterisk (optional)
  6. Interconnect Remote Asterisk Server (optional)

1. Getting Started with Google Voice

With the JS200-FX, you can use any SIP provider including our platinum sponsor, Vitelity. See below for a deal you can’t refuse. But, if you live in the United States, you’d be crazy not to also use Google Voice. It’s free! To use Google Voice with the JS200-FX, you’ll need at least one dedicated Google Voice account. Create a Gmail account first. 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. Then visit http://google.com/voice to set up your Google Voice account and phone number. Yes, you can port an existing number into Google Voice!

IMPORTANT: Do NOT under any circumstances take Google’s bait to switch from Google Chat to Hangouts. Click the X (shown above), or you will forever lose the ability to use Google Chat with your Asterisk appliance. Also 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 Asterisk appliance to work its magic! Otherwise, all inbound and outbound calls will fail. Good News! You’re in luck. Google has apparently had a change of heart on discontinuing Google Chat support so it’s enabled by default in all new Google Voice accounts. Once you’ve created a Gmail and Google Voice account, 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 new voicemails delivered… and transcribed.

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! If you have trouble placing or receiving calls, try BOTH of these tips.

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

2. Connecting to JS200-FX Asterisk Appliance

Now you’re ready to begin the adventure. Turn the switch on the back of the JS200-FX to ON. Plug in the included CAT5 cable between ETH2 port on the JS200-FX and a desktop computer or notebook. Power on the device and wait about 2 minutes. From your computer, browse to 192.168.10.1 and login to Asterisk-GUI as user admin with password for your password. You’ll be prompted to change your password. Make it secure!

3. Configuring Networking on JS200-FX Asterisk Appliance

In a nutshell, you’ll be using the ETH1 port on the JS200-FX to connect to your Internet router. We’ll use ETH2 to directly connect to the JS200-FX from a computer when things go haywire. Assuming your router hands out private IP addresses with DHCP, you don’t really need to do much in the way of network configuration on the JS200-FX unless you want to set up a static IP address for the appliance. You’ll find that option under Networking -> WAN -> Connection Type. We typically recommend permanently assigning the IP address that was handed out by your router within the router’s configuration menu. The real trick at this point is deciphering what that IP address will be. You can figure that out by plugging a CAT5 cable between ETH1 and your router now. The address will appear in the WAN entry under Networking -> Status.

Next, we’ll want to configure the Wireless Networking. We recommend setting the device up as an Access Point under Wireless -> Basic Settings. Under the Wireless Security tab, switch to WPA2-PSK security and create an 8-character password to access the device on its WiFi gateway. This gives you a way to connect wirelessly and be assigned an IP address in the range 192.168.10.100-200. If that range duplicates the private LAN subnet of your router, change it to 192.168.0.

Finally, click on Firewall -> Remote Admin and activate remote access to Asterisk-GUI using port 80. Whatever you do, DO NOT MAP ANY PORTS FROM YOUR FIREWALL TO THIS ASTERISK APPLIANCE! It is an older version of Asterisk that probably is not without some security holes. So long as it’s safely ensconced behind a hardware-based firewall, you should have little to worry about especially if you only use Google Voice trunks for outside calling.

4. Configuring Google Voice on JS200-FX Asterisk Appliance

This is a 5-second task. In the Asterisk-GUI, click Google Voice. Plug in your Google Voice email address and password. If you wish to enable a second Google Voice account, click Enable Line #2 and enter your credentials for the second account. Save your settings and reload the dialplan when prompted. Now plug in a Plain Old Telephone to the TEL1 port on the JS200-FX. To dial out using the first Google Voice account, dial 941 + 1 + the 10-digit number. To retrieve your voicemail, dial 41. For the second Google Voice account, use the 942 prefix and 42 for voicemail.

VoIP 101: Learning the Basics of Asterisk-GUI Management

Everything from here on out is optional reading. But, if you plan to get the most out of your new PBX, you’ve got to master the basics of the lingo so you’ll know how to navigate through and manage the Asterisk-GUI. For the sake of simplicity, we’ll divide calls into three categories: local calls, incoming calls, and outgoing calls. The latter two categories are External calls from or to destinations outside your PBX.

Local Calls. These are Internal Calls between users of your PBX. Users typically are assigned a local phone number, an Extension, on which to receive calls. You connect a telephone to an extension in order to answer and make calls. Traditional analog phones are called POTS phones (a.k.a. Plain Old Telephones). They connect to an FXS port (only!) which is identified by the TEL1 or TEL2 jacks on the JS200-FX. SIP and IAX phones or softphones are digital devices that connect to extensions configured as SIP or IAX extensions/users.

Incoming Calls. As the name implies, these are calls coming into your PBX. You typically rent a phone number (DID) from a Provider. The provider assigns you credentials and registers the DID to a Trunk. On your PBX, you Create and Register a Trunk with credentials matching those assigned by the provider. When a call is placed to your DID, the provider passes the call to your PBX through the registered Trunk. The PBX then identifies both the DID and the CallerID of the incoming call and routes it to a Destination based upon the rules you establish in your Incoming Calling Rules (a.k.a. Inbound Routes). A typical destination would be an Extension or User, a Ring Group or collection of extensions, a Conference Room where multiple callers can converse at the same time, or a Voice Menu (a.k.a. IVR or AutoAttendant).

Outgoing Calls. These are calls destined for Termination on a telephone outside your PBX. It could be across the street or on the other side of the world. Some of these calls are free and some are not. Outgoing calls begin from a Phone connected to an Extension or User. Once a number is dialed, a Dial Plan determines whether the caller is authorized to make the call. If so, the call is passed to the Outgoing Calling Rules (a.k.a. Outbound Routes). These rules determine which Trunk will actually process the call. As with incoming trunks, you sign up for Termination service with a provider that may be the same or different from your DID provider. Outgoing call rules may send calls with a certain Dialing Prefix to a specified Trunk to take advantage of free calling or reduced cost. These calling rules may strip off dialing prefixes and/or add additional digits to the dialed number before it is passed to the Provider for termination on a remote phone.

5. Configuring Asterisk on JS200-FX Asterisk Appliance

Now that you’ve mastered the basics, there’s so much more you can do. In fact, we could write a book about it. Lucky for us (and for you), others have already done that. To get the most out of this terrific appliance, you’ll need to learn more about Asterisk and the Asterisk-GUI. Fortunately, there’s no shortage of tutorials. Start with the JS200-FX Quick Start Guide (PDF). Then take a careful look at Chapter 3 of Digium’s Asterisk Appliance™ Administrator Manual. Next, review Chapter 11 of The Asterisk Book (Second Edition). Finally, review these 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.

6. Interconnecting JS200-FX Asterisk Appliance to Remote Asterisk Server

Interconnecting the new Asterisk appliance to a remote Asterisk server to share outbound trunks or to allow free calls to local extensions on the remote server is easy. First, create an IAX trunk on the remote Asterisk server using a very secure password. This setup will give callers on the Asterisk appliance access to the entire dialplan on the remote Asterisk server so be careful. Also make sure the Trunk Name and username are the same.

On the Asterisk appliance, there are 3 steps: create an IAX trunk to make the connection to the remote server, add an outbound route with a dialing prefix to route calls out the new trunk, and enable the new Trunk in your DefaultLocalContext dialplan.

Trunk setup: Trunks -> New IAX Trunk

You’ll need the IP address or FQDN of your remote server. In addition, the username and password must match what you set up (above) on the remote server.

Outbound Route setup: Outgoing Call Rules -> New Calling Rule

In our example, we’re requiring an 8 prefix followed by a 10-digit number to send a call to the remote server for outbound call processing. If you wanted to force a different dialing prefix at the remote server end in order to send calls out through a specific trunk, that prefix should be Prepended in the highlighted field of the outbound route. This setup would not permit calls to local extensions on the remote PBX. To do that, you’d probably want to create an additional outbound route with a Dial Pattern such as _8XXXX! if the extensions on the remote server were all four digits. Don’t forget to also enable that second outbound route in the dialplan setup below!

Dialplan setup: Dial Plans -> Edit DefaultLocalContexts

Just click on the Out_RentPBX checkbox and Save your update. Then reload the Asterisk dialplan, and you’re all set.

Making Free SIP URI Calls Worldwide

One of the hidden beauties of Asterisk is the ability to place SIP URI calls to anyone in the world and talk for free… for as long as you wish. SIP URIs look much like an email address with a name or number, followed by @, followed by an FQDN or IP address, e.g. 2233435945@sip2sip.info. While the SIP URI setup on the JS200-FX Asterisk Appliance is not exactly straightforward, it’s pretty easy once you know some of Asterisk-GUI’s magic tricks. The simplest method is to Create a New Voice Menu which will work like a Speed Dial for the new SIP URI. For example, here’s the setup to add Lenny to your appliance. Name the new voice menu Lenny and assign a number to the new voice menu (53669 spells L-E-N-N-Y). Now add two Actions by clicking Add New Step twice with the entries shown below. Save your Voice Menu. Then Reload the dialplan. Now dial 53669 to speak to Lenny. Or route telemarketers to this extension as part of your dial plan.

Answer
Macro trunkdial-failover-0.3,sip/2233435945@sip2sip.info,,,

If you’re comfortable using an editor, there’s an easier way using the same methodology included in Incredible PBX for Asterisk-GUI. We’ll actually add a new [CallingRule_SIP_URI] context in which to save SIP URI speed dials. Then we’ll add that new context to the default dialplan: [DLPN_DefaultLocalContexts]. In the future, you can easily add additional SIP URI speed dials to this context. Just give each one a unique extension number and plug in the SIP URI using the syntax shown below.

In the Asterisk-GUI, click Options -> Advanced Options -> Show Advanced Options. Then click on the new File Editor tab. In the Config Files pulldown, choose extensions.conf. Click Add Context button and name it: CallingRule_SIP_URI. The new context will be added to the bottom of the file so go there and click on + to edit its contents. Add the following line and click Save:

exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)

Now we need to add the new context to the default dialplan so search through the contexts until you find [DLPN_DefaultLocalContexts]. Click on the + to edit the context. Then add the following line to the end of the existing list and click Save:

include=CallingRule_SIP_URI

Now click Apply Settings button to save your settings to NVRAM and reload the dialplan. That wasn’t so hard, was it?

There’s another advantage to the second approach. Your Call Detail Records now will actually show the speed dial numbers that are called:

Setting Up Incoming SIP URIs for Your PBX

This is only recommended for those that are highly skilled in Asterisk and those that can afford an expensive phone bill. It requires that UDP port 5060 be exposed to the Internet through your firewall. You need to be extremely careful in setting up SIP URIs to avoid unintended consequences such as allowing strangers to place outbound calls through your PBX on your nickel. The steps are straight-forward. First, configure an FQDN for your server and, if your provider uses dynamic IP addresses, set up dynamic DNS refreshes using the facility included in Networking -> Dynamic DNS. Second, use the File Editor to edit the [general] context in sip.conf. Insert your FQDN into the fromdomain and domain variables. Next, insert the following line: allowexternaldomains=no. Then Save the file. Third, edit the [default-public] context in extensions.conf. Insert your desired SIP URIs in this context using the proper syntax. For example, to route a SIP URI for mothership@FQDN.yourdomain.com to extension 6001, the dialplan code would look like this: exten=mothership,1,Goto(default,6001,1). To route the same SIP URI to your first Voice Menu, the code would look like this: exten=mothership,1,Goto(voicemenu-custom-1,s,1). To route the same SIP URI to your first Ring Group, use: exten=mothership,1,Goto(ringroups-custom-1,s,1). To route the incoming SIP URI to an outgoing SIP URI, use: exten=mothership,1,Dial(SIP/somewhere@someFQDN.somedomain.com).

There’s a silver lining to activating an inbound SIP URI. Once it’s properly configured, you can sign up for a free phone number in the Seattle area and map that DID to the SIP URI of your server. All of the incoming calls are free! This gives you some redundancy in the event of a Google Voice outage. Just visit www.ipkall.com to sign up for your free number.

Hardening the JS200-FX Firewall

Particularly if you elect to support incoming SIP URIs, you’ll want to tighten up the SPI Firewall included in the JS200-FX. While we have no simple way to decipher the existing rules, you can add rules of your own to lessen the opportunity for mischief. This is especially important in the SIP arena. Just to be sure you don’t lock yourself out of your own server, we recommend a 4-step process: (1) allowing full access from private LAN subnets, (2) whitelisting the FQDNs and IP addresses from which you will access the JS200-FX, (3) whitelisting the providers that you intend to use as well as the IP addresses of external phone devices, and (4) locking down incoming SIP URI access to a single FQDN for your server. The fourth step keeps random strangers from attempting to gain SIP access by scanning blocks of IP addresses in search of vulnerable servers. It’s a good idea to use an obscure FQDN for your appliance which minimizes the ability of strangers to guess the acceptable SIP URIs, e.g. somefunkyFQDN.somedomain.net would block all incoming SIP URI attempts by either IP address or by guessing any other FQDN. In other words, the FQDN works just like a password. Thus, if you set up a mothership SIP URI (make up your own!), the only incoming SIP URI calls that would be allowed would be those calling mothership@somefunkyFQDN.somedomain.net. Don’t publish the actual SIP URI anywhere!

Also be advised that, if you use FQDNs in the step #2 white list and the dynamic IP address of these FQDNs changes, you will need to manually restart the JS200-FX to enable the new IP address. Currently, there is no ability to check for FQDN changes and automatically restart the appliance.

To create the new firewall rules, choose Firewall -> Custom Rules -> Enable ON. Then enter and SAVE & APPLY the following rules using your actual settings rather than the sample entries below. CAUTION: This data should be entered by accessing the JS200-FX via WiFi at the 192.168.10.1 address, or you may lock yourself out during the update process.

#1 private subnets and loopback - no changes needed in this section
-A INPUT -s 192.168.0.0/16 -j ACCEPT
-A INPUT -s 10.0.0.0/8 -j ACCEPT
-A INPUT -s 172.16.0.0/12 -j ACCEPT
-A INPUT -s 127.0.0.0/8 -j ACCEPT

#2 enter your own IP addresses for WhiteList access below
-A INPUT -s homeFQDN.dyndns.org -j ACCEPT
-A INPUT -s alternateFQDN.dyndns.org -j ACCEPT
-A INPUT -s 129.43.13.220 -j ACCEPT

#3 providers and interconnected servers and phone devices
## atlanta.voip.ms sample entry
-A INPUT -s 174.34.146.162/32 -p udp -m multiport --dports 5060,5061,5062,5063,5064,5065,5066,5067,5068,5069,5080,4569 -j ACCEPT

#4 SIP URI access - enter JS200-FX FQDN in next line and leave the rest
-A INPUT -p udp --dport 5060 -m string --string "REGISTER sip:somefunkyFQDN.somedomain.net" --algo bm -j ACCEPT
-A INPUT -p udp --dport 5060 -m string --string "REGISTER sip:" --algo bm -j DROP
-A INPUT -p udp --dport 5060 -m string --string "OPTIONS sip:" --algo bm -j DROP
-A INPUT -p udp --dport 5060 -j ACCEPT

Implementing 7-Digit Dialing with Your Favorite Area Code

Once you have at least one Google Voice account set up, here’s another trick to implement 7-digit dialing with your favorite area code. Just add an additional line to the [CallingRule_SIP_URI] context substituting your area code for 843:

exten=_NXXXXXX!,1,Macro(trunkdial-failover-0.3,${GoogleVoice_1}/1843${EXTEN},,GoogleVoice_1,)

OK, Smarty Pants: Show Me the Beef!

We know what some of you are thinking. “Do you really know as much about VoIP as Lenny does?? Before wasting 30 bucks 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. Then press 1. You’ll be connected to the Conferencing System running on the JS200-FX Asterisk Appliance. You can chat with other Nerd Vittles users that have joined before you. So… what are you waiting for? Click away and try the JS200-FX Appliance for yourself.



You can implement this Click-to-Dial technology using your own JS200-FX Asterisk Appliance in about 10 seconds. Once you have configured Google Voice as outlined in Step #1 above, click on the Call Widgets tab under Settings. Click Add a New Call Widget, give it a name, turn off ringing your home or office phone, turn off Call Presentation, and Save Changes. Now simply cut-and-paste the Embed code that’s provided and insert it into a public web page of your choice. Doesn’t get much easier than that, and all your family and friends can call you for free from the convenience of any available telephone in the U.S. or Canada by simply clicking on the Call Me widget on your web site’s home page.

Originally published: Monday, March 16, 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…

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