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It’s All About Choices: Incredible PBX ISO Now Available in the Vultr VPS Cloud
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We’ve heard you. Many prefer the ease of installation offered by the Incredible PBX ISO. Unfortunately, until now, it wasn’t available if you wanted a do-it-yourself Cloud Solution other than, of course, using a turnkey Incredible PBX server at RentPBX. Well, now you have a choice. A great new VPS provider, Vultr, now gives you the option of using your own ISO when you deploy Cloud servers on their worldwide platforms. It’s not 100% intuitive just yet, but Vultr will get there. In the meantime, we’ll walk you through the procedure if you’d like to try it yourself. The good news is Incredible PBX can run on a $5/month Vultr server with 768MB of RAM plus a 768MB cache and a 15GB SSD plus 1TB of monthly bandwidth. And you can build your server in less than 30 minutes.
Getting Started with VULTR
Before you can set up virtual machines at VULTR, you’ll obviously need an account. Help Nerd Vittles keep the lights on by using our referral link. It doesn’t cost you a nickel extra. In fact,for a limited time, you get a $20 credit to try out the service when you sign up using our link above.
Once you’ve set up your account and logged in, the first step is to upload the Incredible PBX ISO so that you can use it to build cloud-based servers:
(1) Click on the Servers link in the left column.
(2) Click the ISO tab.
(3) Click the Add ISO button and use the following Incredible PBX ISO link to upload the ISO to VULTR:
http://sourceforge.net/projects/pbxinaflash/files/IncrediblePBX13.2.iso
(4) Once the ISO has been uploaded, it will display in your ISO uploads listing as shown above.
Building an Incredible PBX Server at VULTR
Now you have two ways to build Incredible PBX servers at VULTR. You can do it the traditional way by starting with a 64-bit CentOS 6.7 image and running the installer. Or you can use your new Incredible PBX ISO which we’ve preloaded into your VULTR account. Let’s do it the new way just to document the procedure.
Start by clicking on the Servers link in the left column. Then click + Deploy New Server button.
(1) Choose a Server Location from the list of choices shown above. For PBXs it’s always good to choose a location that’s near to both your VoIP hosting provider(s) and your actual phones where possible.
(2) For Server Type, click on the Custom ISO tab as shown above and select IncrediblePBX13.2.iso.
(3) For Server Size, choose the platform that meets your needs. You can always start small, take a snapshot of your server, and rebuild with a larger platform down the road if your requirements change.
(4) Choose any Additional Features and HostName desired. Then click the Deploy Now button.
(5) Click the Servers link to monitor the progress in building your virtual machine.
Installing Incredible PBX at VULTR from the ISO
Once your virtual machine Status changes from Installing to Running, (1) write down the IP address of your new server and click Restart button. (2) Click Manage. (3) Click View Console. Choose the type of Incredible PBX install you prefer and follow the prompts to enter your root password and time zone. When the operating system phase of the install finishes, your virtual machine will reboot. While this phase is underway, you’ll need to get prepared to disable the ISO during the reboot. You don’t want to inadvertently start the install all over again!
While the Phase I install is underway, click Server Details in your browser. Then choose (1) Settings, (2) Custom ISO, and (3) click Remove ISO. WAIT at the Are You Sure prompt until the reboot begins in the Console Window. Then click YES in your browser and also close the Console window.
Now use SSH or Putty to login to your new server as root at the IP address you wrote down above. Once logged in, run the Incredible PBX installer to begin Phase 2 of the install: ./IncrediblePBX13-12R.sh. Choose the type passwords you wish to use for Google Voice, and the install will begin. In order to bring your server up to current specs, it will reboot once more after a few minutes. Then you’ll need to kick off the install once again: ./IncrediblePBX13-12R.sh. Choose the type passwords you wish to use for Google Voice, and the install will finish without further interruption. When the install finishes, the server will reboot to bring Asterisk and your firewall on line.
Login to your server as root one more time to get the latest Incredible PBX updates and set up your passwords and verify your time zone. Then you’ll be ready to handle future server admin using your browser:
Make your root password very secure: passwd Create admin password for GUI access: /root/admin-pw-change Create admin password for web apps: htpasswd /etc/pbx/wwwpasswd admin Set your correct time zone: /root/timezone-setup Make a copy of your Knock codes: cat /root/knock.FAQ Review current info about your server: status
If you want to add fax support to your PBX, we’ve got one more installer to run, and you’ll be all set. Just run the following command and press the ENTER key each time you are prompted for input (about 21 times):
cd /root ./incrediblefax11.sh
Success!
It’s a GUI, GUI World Out There
Most management of Incredible PBX will be handled using your favorite browser and one of two GUIs: the Kennonsoft menu which appears when you access the IP address of your server using a browser AND the Incredible GUI which is available on the Admin page of the Kennonsoft menu.
Choose Incredible GUI Administration from the Admin menu of the Kennonsoft GUI (shown above) by clicking on User to switch. The default username is admin and the password is what you set in the preceding step. Now edit extension 701 so you can decipher or change the randomized passwords that was assigned to the 701 extension and its voicemail account: Applications -> Extensions -> 701.
Setting Up a Soft Phone to Use with Incredible PBX
Now you’re ready to set up a telephone so that you can play with Incredible PBX. We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension
701 password. Choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of your server, 701 for your account name, and whatever password you created for the extension. Click OK.
Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:
DEMO - Allison's IVR Demo
947 - Weather by ZIP Code
951 - Yahoo News
*61 - Time of Day
*68 - Wakeup Call
TODAY - Today in History
Now you’re ready to connect to the telephones in the rest of the world. If you live in the U.S., the easiest way (at least for now) is to set up a free Google Voice account. Google has threatened to shut this down but as this is written, it still works. An update supporting Google Voice OAUTH 2 authentication is available here with a complete tutorial. Just start at step #1b. A better long-term solution is to choose several SIP providers and set up redundant trunks for your incoming and outbound calls. The PIAF Forum includes dozens of recommendations to get you started. You’ll also find an incredible deal from our platinum sponsor, Vitelity, at the end of this article.
Configuring Google Voice
If you want to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX. If you want to use the inbound fax capabilities of Incredible Fax, then you’ll need an additional Google Voice line that can be routed to the FAX custom destination using the GUI. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!
We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX. It’s free at least through 2013. Google Voice no longer is by invitation only so, if you’re in the U.S. or have a friend that is, head over to the Google Voice site and register.
You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it’s over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…
IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.
While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:
- Call Screening – OFF
- Call Presentation – OFF
- Caller ID (In) – Display Caller’s Number
- Caller ID (Out) – Don’t Change Anything
- Do Not Disturb – OFF
- Call Options (Enable Recording) – OFF
- Global Spam Filtering – ON
Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.
One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!
Now you’re ready to set up your Google Voice trunk in the GUI. After logging in with your browser, click the Connectivity tab and choose Google Voice/Motif. To Add a new Google Voice account, just fill out the form. Do NOT check the third box or incoming calls will never ring!
IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.
If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
Troubleshooting Audio and DTMF Problems
You can avoid one-way audio on calls and touchtones that don’t work with these simple settings in the GUI: Settings -> Asterisk SIP Settings. Just plug in your public IP address and your private IP subnet. Then set ULAW as the only Audio Codec.
A Few Words about the Incredible PBX Security Model for CentOS
Incredible PBX for CentOS joins our previous Ubuntu build as our most secure turnkey PBX implementation. As configured, it is protected by both Fail2Ban and a hardened configuration of the IPtables Linux firewall. The latest release also includes Port Knocker for simple, secure access from any remote computer or smartphone. You can get up to speed on how the technology works by reading the Nerd Vittles tutorial. Your Port Knocker credentials are stored in /root/knock.FAQ together with activation instructions for your server and mobile devices. The NeoRouter VPN client also is included for rock-solid, secure connectivity to remote users. Read our previous tutorial for setup instructions. As configured, nobody can access your PBX without your credentials AND an IP address that is either on your private network or that matches the IP address of your server or the PC from which you installed Incredible PBX. You can whitelist additional IP addresses by running the command-line utility /root/add-ip. You can remove whitelisted IP addresses by running /root/del-acct. Incredible PBX is preconfigured to let you connect to many of the leading SIP hosting providers without additional firewall tweaking. We always recommend you also add an extra layer of protection by running your server behind a hardware-based firewall with no Internet port exposure, but that’s your call. And it’s your phone bill. 😉
The IPtables firewall is a complex piece of software. If you need assistance with configuring it, visit the PIAF Forum for some friendly assistance.
Incredible Backup and Restore
We’re also pleased to offer our latest backup and restore utilities for Incredible PBX. Running /root/incrediblebackup will create a backup image of your server in /tmp. This backup image then can be copied to any other medium desired for storage. To restore it to another Incredible PBX server, simply copy the image to a server running Asterisk 13 and the same version of the Incredible PBX GUI. Then run /root/incrediblerestore. Doesn’t get much simpler than that.
Incredible PBX Automatic Update Utility
Every time you log into your server as root, Incredible PBX will ping the IncrediblePBX.com web site to determine whether one or more updates are available to bring your server up to current specs. We recommend you log in at least once a week just in case some new security vulnerability should come along.
In the meantime, we encourage you to sign up for an account on the PIAF Forum and join the discussion. In addition to providing first-class, free support, we think you’ll enjoy the camaraderie.
Incredible PBX Application User’s Guide
And now would be a great time to learn about the dozens of Incredible PBX applications. Enjoy!
Originally published: Tuesday, July 12, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
2016: The Year of the May Bromance with XiVO, Asterisk 13, and the GPL
Remember that girl that you never much noticed in high school that happened to cross your path years later and there was that instant attraction. It’s one of our favorite country songs.
What a month it has been. 30 days ago we’d never met. And then it happened. We were introduced to XiVO by a single post on the PIAF Forum. It’s difficult to choose a single adjective that describes how feature-rich XiVO is as a real-time Asterisk platform. Ten years of development coupled with some really talented Canadian developers puts XiVO in a league of its own. Pure GPL code. No gotchas. No hidden agenda. Incredible documentation. No snake oil salesmen trying to peddle add-ons for money. And, thanks to the endless patience of Sylvain Boily and Pascal Cadotte Michaud, two of the key XiVO developers, we’ve had quite a joy ride this month. Today we want to share what we’ve learned and provide everything you need step-by-step to hit the ground running with XiVO. In less than an hour, you’ll be sitting in front of your own XiVO server, and you can judge for yourself. All we can ask is "Where have you been all these years?" XiVO really is THAT GOOD!
UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article superseded by the new release.
Let’s Start with the Features. If you make frequent changes to your Asterisk platform, then you’ll really appreciate XiVO’s realtime implementation. Changes are loaded almost instantaneously. Contrast that with some other platforms with hundreds of users where simple changes require several minutes or more to reload the Asterisk dialplan. Here’s an example:
Speaking of hundreds of users, consider the time required to generate accounts and phone configurations for hundreds of users. With XiVO, a simple spreadsheet can be used to build all the user accounts in seconds. And XiVO’s Endpoint Manager supports all of the major phone manufacturers and configures your phones in seconds. With other solutions, you’ve got a fee for the Endpoint Manager add-on itself and then another annual maintenance fee to assure that the software will continue to work. Contrast that with XIVO’s GPL alternative. Choosing Configuration:Provisioning:Plugins:Update generates templates for dozens of great SIP phones in seconds with just four button clicks. Adding your own new models is a breeze.
We’ve already written about XiVO upgrades and backups, the twin-edged sword with most Asterisk implementations. Yes, you can go the proprietary route and stumble through dozens of menus and arcane commands to load upgrades and create backups. With XiVO, backups are automatic and the updates are fast and furious. Every morning a new backup arrives in /var/backups/xivo, and a new version of XiVO is released every three weeks. You can upgrade your server in under a minute with one simple command. Did we mention reported bugs are addressed in under 30 days!
One of the real failures of the other GUI offerings is their lack of tools to perform most tasks in any way other than using the GUI. For developers and those that maintain numerous phone systems from afar, this becomes a nightmare. XiVO offers a better alternative. Yes, there’s an incredible GUI. But there’s also a robust collection of APIs that provide programmatic access to manage and query virtually every piece of the XiVO puzzle. As if that weren’t enough, there’s also real documentation and samples to show you how to use each piece.
Have we got your attention yet? We haven’t even touched upon Call Centers and High Availability (HA) server deployments, but they’re included at zero cost. For today and just getting started, suffice it to say that XiVO covers all the bases with open source code. And all of these features already are incorporated into the product you’re about to install. So, if those two items have been on your Wish List and you’d prefer to avoid paying an arm and a leg for proprietary, commercial add-ons with recurring annual fees, today’s your lucky day. Break open the latest XiVO documentation, all 479 pages of it and enjoy!
We’ve barely scratched the surface of what you can do with XiVO. Simply stated, anything you can do with the other GUI offerings, you can do better, quicker, and cheaper with XiVO. And, if there are features you need that aren’t there, all you have to do is ask. We’re porting applications to XiVO at the rate of about one new application a day. You can do the math on feature sets and measure where we’ll be when the summer is over.
Getting Started with XiVO. Jump to the latest tutorial to install XiVO.
Bookmark Getting Started with XiVO and you can follow our progress in coming months. Enjoy!
For additional tips & tricks with XiVO, take a look at our previous articles:
2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business
The XiVO Adventure Continues: Adding Incredible PBX Goodies to Your Sandbox
2016, Celebrating The Preakness: CallerID Superfecta Rides Again with XiVO
Choosing Wisely: Mastering Asterisk IVR and AutoAttendant Design with XiVO
Choosing a XiVO Hardware Platform
- XiVO Platform Tutorial: Installing XiVO on a Dedicated PC
- XiVO Platform Tutorial: Installing XiVO VM on VirtualBox
- XiVO Platform Tutorial: Installing XiVO VM on VMware ESXi
- XiVO Platform Tutorial: Installing XiVO VM at ImpactVPS
- XiVO Platform Tutorial: Installing XiVO VM at Digital Ocean
- XiVO Platform Tutorial: Installing XiVO VM at CloudAtCost
- XiVO Platform Tutorial: Installing XiVO VM at RentPBX
XIVO Initial Setup Tutorial
XiVO Initial Setup Tutorial: Getting Started with XiVO
XIVO Trunk Implementation Tutorials
- XiVO Trunks Tutorial: Installing a Vitelity SIP Trunk
- XiVO Trunks Tutorial: Installing a VoIP.ms SIP Trunk
- XiVO Trunks Tutorial: Installing a RingPlus SIP Trunk
- XiVO Trunks Tutorial: Installing a FreeVoipDeal (Betamax) SIP Trunk
- XiVO Trunks Tutorial: Installing a Google Voice-Simonics SIP Trunk
- XiVO Trunks Tutorial: Installing an Anveo Direct Outbound SIP Trunk
- XiVO Trunks Tutorial: Installing a Skype Connect SIP Trunk
XIVO Call Routing Tutorials
- XiVO Call Routing Tutorial: Creating Inbound Routes for DIDs
- XiVO Call Routing Tutorial: Creating Outbound Routes for PSTN Calling
Taking Nerd Vittles’ XiVO IVR for a Test Drive
There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:
Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)
Published: Tuesday, May 31, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Mastering XiVO IVR and AutoAttendant Design
Today we want to talk a little about design choices and IVRs. First and foremost, we don’t want to leave anyone behind during our XiVO adventure. XiVO is a platform that adjectives really can’t describe. It’s that good and, frankly, we’re having a hard time believing it’s been around for almost a decade and nobody much talked about it. Leave it to the crazy Americans to only look at stuff from the U.S. of A. Funny thing is that the two major GUIs for Asterisk® now are both Canadian-based.
America calling pic.twitter.com/bMqg601m8l
— 🇺🇦Evan Kirstel #B2B #TechFluencer (@EvanKirstel) May 24, 2016
One of our PIAF Forum readers posted a comment last week that said:
The only downside I see is that XiVO does not have [a] GUI for building IVRs. To build [a] complex, nested IVR system, everything has to be thought about in great detail writing contexts and dial plans to suit your unique requirements. It would be nice if XiVO offered a GUI for building IVRs.
This raises some issues about GUI design and development that are worth addressing. As with any GUI, the development cycle is lengthy and incredibly complex. This is especially true with XiVO where new versions are released every two weeks! In our second XiVO article, we showed how easy the upgrade procedure was. Those coming from other Asterisk platforms will appreciate this little shocker. XiVO doesn’t break stuff with their upgrades. Frankly, the only other company I can say that about is SONOS. If you don’t have their music platform, you’re missing a treat.
Introducing new components into any "main product" can cause all sorts of problems with the pieces that used to work. If you don’t believe it, look at some of the "other forums" and look at the number of message threads complaining that the new X Widget broke the Y widget and now nothing works. While we can’t speak for everyone, I think it’s safe to say nobody that depends upon their phone system wants to see it go up in flames regularly because some developer had a great new idea that didn’t quite do what it was supposed to do.
To their credit, the XiVO developers were smarter than that. They’ve not only built a mighty mousetrap, but they’ve done it in a way that supports outside integration of additional components without breaking the main product. There are numerous "hooks" that allow anyone with any skill set to add missing pieces. Some of these hooks are exclusively for programmers, but many were designed to let anybody integrate almost anything into the XiVO platform.
So, when a user says "I wish XiVO had an IVR Builder in the GUI," our first inclination was to chuckle and respond with "You just don’t appreciate how lucky you are not to have an IVR Builder in the GUI." What the commenter didn’t appreciate is that you don’t need to pre-build components with XiVO before developing an IVR. With the "other" GUI, you first had to create Custom Destinations and Custom Contexts and Miscellaneous Applications and Miscellaneous Destinations and Custom Recordings in the GUI before you could take advantage of the IVR GUI to build much of anything. Think about that for a minute. Yes, there was an IVR builder but, before you could use it, you first had to transform every component to be incorporated into the IVR using a large number of subcomponents to translate all of your Asterisk pieces into the GUI’s special lingo. Think of them as GUI pigeonholes, and you had to decipher which Asterisk square pegs went in which GUI round holes. We can’t count the number of times we’ve begun the IVR creation process only to have to stop and create missing components because the IVR builder simply wouldn’t recognize a feature as being part of our Asterisk dialplan.
Building IVRs and AutoAttendants with XiVO
The anatomy of an IVR in Asterisk could not be more straight-forward. You have a prerecorded message that plays to the caller giving them choices from which to choose from a menu of selections. The caller presses one of the 12 keys on their phone, and the IVR goes off and does some task: calls an extension, plays a recording, runs an Asterisk application, makes an outside call, or kicks off another IVR with another recording and more choices. Some options in the IVR may not be mentioned, and this is commonly referred to as the Stealth AutoAttendant. None of this is rocket science.
To build an IVR, you need these components: (1) a prerecorded message, (2) a list of the choices you want to provide to the caller with the corresponding destinations on the PBX to execute those choices, and (3) a template to follow to create the IVR dialplan code in XiVO.
Trust us when we say the major problem with IVRs is not that they’re difficult to build in XiVO. The real issue with most IVRs is that the person that implemented the IVR spent all their time worrying about the mechanics of PBX implementation and didn’t put sufficient thought into the IVR layout and the caller’s experience when actually interacting with the IVR. If you haven’t heard Allison Smith speak about IVR design, put it on your Bucket List for the next AstriCon or do some reading. That’s a long-winded way of saying that filling in the blanks of an IVR template is just as easy as point-and-click or drag-and-drop except for the eye candy. Just be thankful the XiVO platform gives you the flexibility to do it yourself without having to create imaginary destination hooks and recording linkages before they can be used in the product’s IVR GUI because the developers didn’t have the foresight to think outside their own GUI’s box. Every Windows user can appreciate that problem.
For today, we’re assuming you’ve done your homework and have already sketched out the options you want to incorporate into your IVR or IVRs. No GUI can help with that! So we’ll pick up from there and show you how easy it is to incorporate your IVR design into XiVO.
Adding Prerecorded Messages in XiVO
For openers, you obviously need a recording to greet callers and tell them what their choices are when using your IVR or AutoAttendant. You can build these recordings yourself on the XiVO platform or, for a more professional IVR, you can send the text off to Allison Smith and let her record the voice prompts for you. Digium makes it easy. Visit their web site, type in the text, and you’ll have your recording in a couple of days. No, they’re not free, but they’re not expensive either.
Since we’re just getting started, let’s assume you want to create a recording prototype on your own to work out the kinks in your IVR first. Here’s how. We’re assuming you’ve already read the Nerd Vittles XiVO tutorial and put the Festival TTS platform in place. Next, log into your XiVO server as root. To keep things simple, let’s put the recordings in WAV format in the /var/lib/xivo/sounds/playback directory which is reserved for our custom recordings:
cd /var/lib/xivo/sounds/playback
To actually generate the sound file that Asterisk can play back, execute the command below after placing your text between the quotation marks and giving the sound file a name, e.g. ivr-number1.wav:
echo "Text goes here" | /usr/bin/text2wave -F 8000 -o ivr-number1.wav
Here’s an example:
echo "Thank you for calling. Press 1 for Tom, 2 for Dick, or 3 for Harry. Press 0 to be connected to the operator." | /usr/bin/text2wave -F 8000 -o ivr-number1.wav
Marrying IVR Choices to PBX Destinations
Whether you’re deploying an IVR using FreePBX® or XiVO, you still have to translate your Plain English options into code that the GUI understands so that calls get routed successfully to the intended destinations.
Let’s begin with the FreePBX Way. Our previous IVR tutorial showed how it was done:
As you can see from the above routing procedure, there were interim steps for every single option in this IVR menu except #8. What you may not appreciate is that you first had to create both a Misc Destination AND a Custom Extension before these options could be used in FreePBX. Otherwise, the options simply didn’t appear in the IVR GUI’s pull-down pick lists.
If you wished to incorporate a custom context that wasn’t assigned an extension number on your PBX, there was a different GUI procedure. For something as simple as retrieving the time of day, you had to get the custom context registered with FreePBX before the dialplan code could be used in the IVR. According to the FreePBX developers, this functionality was considered an "advanced feature and should only be used by knowledgeable users."
Our purpose in documenting all of this is to demonstrate that building IVRs even in a GUI is much more than point-and-click. It requires mastery of some fundamental Asterisk dialplan concepts not to mention the GUI’s own labyrinth of secret pigeonholes. Once you’ve had to master all of that, we believe it’s simpler to build IVRs using simple commands rather than jumping through all of the convoluted hoops required just to make your IVR GUI platform happy.
Let’s compare this methodology to the XiVO way of doing things by way of example. Then you can decide for yourself which approach is more complex. Would you know all of these on your own? Probably not. But now you can see how simple it really is. There really are only two words you need to learn: Dial and Goto. 🙂
Call an Extension: Dial(Local/701@default) Call a Ring Group: Dial(Local/801@default) Call a PSTN Number: Dial(Local/8005551212@default) Call a SIP URI: Dial(SIP/2233435945@rentpbx.mundy.org) Access DISA with permission: Dial(Local/3472@default) Join a Conference: Dial(Local/2663@default) Playback Yahoo News: Dial(Local/951@default) Playback Weather Forecast: Dial(Local/947@default) Identify IVR Option as Invalid and Repeat Menu: Goto(i,1) Hangup on Caller for Choosing Invalid Option: Goto(t,1) Execute Time of Day Custom Context: Goto(new-time,s,1) Send Caller to a Second IVR and Play Second Recording: Goto(ivr-2,s,3)
Building XiVO IVRs from an IVR Template
We can’t speak for everyone, but we’ve always told folks not to write a book about how to do something. Just give us an example that’s easy to follow and we’ll take it from there. So here you go.
In the XiVO world, IVRs are nothing more than custom contexts. They have a name in [brackets], and they’re stored in config files saved in /etc/asterisk/extensions_extra.d. A config file can include multiple contexts or only one. For IVRs, we recommend you save each one in a single configuration file that houses a single context.
We’re going to give you a template to follow in creating all of the IVRs you can dream up. All you need is a custom recording for each one and your list of choices and destinations for those choices. The examples above tell you everything you need to know to build awesome IVRs.
After downloading the template, we recommend that you not edit it directly. Make a copy with a new file name and change the context name in the template to match your new file name. We also do one other little trick with all of our custom contexts. They always begin and end with comment lines like this using the context name:
;# // BEGIN ivr-template ;# // END ivr-template
The reason for this is it makes it incredibly easy to remove the entire context with a single command:
sed -i '\\:// BEGIN ivr-template:,\\:// END ivr-template:d' ivr-template.conf
This doesn’t matter so much when you only have a single context in a single file. But it is immensely helpful when you’ve stored dozens of contexts within the same file. Some may prefer to store all of the related IVR contexts for their entire IVR tree in a single file. And then you’ll appreciate this tip when it’s time to make major changes in your IVR.
Let’s begin by putting your template in place and then cloning it to ivr-number1:
cd /etc/asterisk/extensions_extra.d wget http://incrediblepbx.com/ivr-template.tar.gz tar zxvf ivr-template.tar.gz rm -f ivr-template.tar.gz cp -p ivr-template.conf ivr-number1.conf sed -i 's|ivr-template|ivr-number1|' ivr-number1.conf
The rest of today’s exercise can be performed in the XiVO GUI using its built-in editor. Open the GUI with your browser and navigate to Services -> iPBX -> Configuration files and then open ivr-number1.conf by clicking on the pencil icon beside it.
Anatomy of the XiVO IVR Template
First things first. Change the sound recording in line s,3 to match the recording you made above without the .wav extension: ivr-number1. Leave the directory path just as it is. So your line should now look like this:
exten => s,3(skip),Set(IVR_MSG=/var/lib/xivo/sounds/playback/ivr-number1)
Next, take a look at the structure of the file. You’ll note that there are options labeled exten => 0,1, through exten => 9,1,. These match the numeric keys on a telephone obviously. In the IVR world, it’s called a phone tree. All you need to change is what comes after the second comma on each line. This destination should be one of the XiVO commands we documented above telling XiVO how to process the call. For option 0, let’s assume you wanted to route the call to extension 701. Your 0 branch would look like this:
exten => 0,1,Dial(Local/701@default)
The remaining dial options should be obvious. If you want to designate a particular option to be invalid, make the option look like this:
exten => 9,1,Goto(i,1)
Another alternative is to remove the line entirely; however, we prefer the above approach because it makes it easy to change things down the road if you decide to use option 9 as a call destination.
Two other options warrant a brief explanation. The i option tells XiVO how to process the call if the caller chooses an invalid option. The t option tells XiVO what to do if the 3-second timeout occurs without the caller pressing a key. You can modify these to meet your own requirements. As configured, an invalid option sends the caller back to the recording to start over. And the timeout option hangs up the call.
Finally, phone trees can get quite complex. A GUI can’t fix that either. Pressing option 2 might trigger phone tree 2 while pressing 3 might trigger phone tree 3. Programmers could obviously rewrite the dialplan to handle all of these separate phone trees with their separate branches in one giant, convoluted chunk of dialplan code. But why? Just make each phone tree a separate IVR housed in its own file with its own context. And navigate between the IVRs using simple Goto commands such as Goto(ivr-number2,s,3). To return to the main IVR, do the same thing pointing to the line number to which the call should be redirected, e.g. Goto(ivr-number1,s,3). You obviously don’t need to answer each call but once so skip those lines in the IVR dialplan when choosing the line number to which to redirect processing.
Routing Incoming Calls to Your IVR
If you’ve already set up one or more DIDs on your PBX, then you probably routed those Incoming Calls to a user or ring group. Changing the routing to send the calls to your IVR is easy. Just edit the DID entry for the Incoming Calls you wish to redirect and set the Destination to Customized and the destination Command to the context of your IVR: Goto(ivr-number1,s,1). Save your change and you’re all set. Remember, XiVO is a real-time Asterisk server so all of your changes take effect immediately. There’s no rewriting of the entire Asterisk dialplan. Enjoy!
Letting Callers Dial Extensions Within IVR
Some administrators prefer to let callers dial an extension directly while an IVR is playing. You can easily add this functionality in XiVO. This post on the PIAF Forum showed how. Simply edit /etc/asterisk/extensions_extra.d/ivr-1.conf and modify the code like this. Be sure to change the number of X’s in the last line to match the length of your extension numbers. Then reload your dialplan.
;exten => s,n,ExecIf($["${IVR_MSG}" != ""]?Background(${IVR_MSG})) ;exten => s,n,WaitExten(10,) exten => s,n,Read(Digits,${IVR_MSG}) exten => s,n,Goto(${Digits},1) exten => _XXX,1,Dial(Local/${Digits}@default)
Taking Nerd Vittles’ XiVO IVR for a Test Drive
There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:
Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)
Published: Thursday, May 26, 2016
UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article as a supplement to the new release.
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
2016, Celebrating The Preakness: CallerID Superfecta Rides Again with XiVO
If you missed The Preakness Saturday, another Triple Crown bit the dust… or the mud in this case. But, if you had the winning Superfecta ticket, you made a $316 profit on your $1 bet. For the rest of us, there’s still a Superfecta win to celebrate, and this one’s free. We’ve begun porting CallerID Superfecta to the XiVO platform and today we’ll share that code with you together with lots of other goodies in our third roundup of Incredible PBX add-ons for the XiVO PBX. If you’re just joining the party, start with the first and second articles on XiVO, and then you’ll be ready to roll up your sleeves for Chapter 3.
UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article superseded by the new release.
Installing CallerID Superfecta for XiVO
As we mentioned in April, it’s always nice to see your baby grow up. Nearly a decade ago, we introduced an AGI script for Asterisk@Home known as CallerID Trifecta for FreePBX® 2.2.0. As sources of CNAM lookups expanded, a number of other individuals contributed code to support those lookups. When we added a fourth CNAM lookup source, the original application morphed into CallerID Superfecta. Then we gave up. The source lookups became too numerous to mention.
For today, we’ve changed the design a bit to better accommodate the XiVO platform. There’s a single AGI script that houses the various CNAM lookup sources and the code to extract CallerID names from those sources. And there’s a dialplan script that let’s you specify which CNAM sources to use and in which order. As with the original release, CallerID lookups take the phone number of the caller and walk through your CNAM lookup sources in the order you specify until a CallerID name match is found. Then the result is returned to the PBX for use with the incoming call. The reason for all of this is historical. The Bell Sisters decided it was more profitable to dump CallerID name information in the bit bucket rather than passing it along with incoming calls. In that way, they could charge folks for looking up the matching name in their proprietary databases. A few CallerID lookup sources remain free, but many now are pay-as-you-go platforms with a typical lookup costing about half a cent. Unfortunately, all providers consider "WIRELESS CALLER" a successful lookup. Ka-Ching! We’ve documented the procedure to add additional CNAM lookup sources on the PIAF Forum. Please share your work!
This release of CallerID Superfecta provides four lookup sources. That’s what a Superfecta is all about, picking four winners:
0 - AsteriDex SQLite3 database
1 - OpenCNAM (free from cache or commercial)
2 - BulkCNAM (commercial only with free trial)
3 - TelcoData (provider, city, and state of caller)
There are three simple steps to putting everything in place. First, run the scripted commands below. Second, specify which CNAM sources you wish to use and in what order. Third, register with the commercial providers you’d like to use and plug your credentials into the CallerID Superfecta script.
To install CallerID Superfecta, log into your server as root and issue the following commands:
cd / apt-get -y install php5-xmlrpc wget http://incrediblepbx.com/cid-superfecta.tar.gz tar zxvf cid-superfecta.tar.gz rm -f cid-superfecta.tar.gz /etc/init.d/asterisk restart
By default, CallerID Superfecta will attempt to use all four of the providers in the order shown to retrieve a CNAM match. If you have migrated your AsteriDex database to XiVO as we covered in last week’s article, then CallerID names will be provided for your most frequent incoming calls without ever accessing external sources. You won’t break anything by leaving all four CNAM sources activated. But, without signing up for service with OpenCNAM or BulkCNAM, your CNAM results will be diminished considerably. And a result of "WIRELESS CHARLESTON SC" from TelcoData doesn’t provide much of a clue as to who is calling. But at least you don’t get charged for that one.
In the next release, we will add an optional feature that will populate entries in AsteriDex from CNAM data returned from OpenCNAM and BulkCNAM. The good news is, if you leave AsteriDex at the top of the CallerID Superfecta search list, you’ll never pay for the CNAM lookup of the same number twice. The bad news is, to keep the bad guys from self-populating your database with expensive phone numbers, you’ll need to password-protect the Voice Dialing application if it is part of your inbound IVR.
To change the source list or sequence of CNAM lookups, open the XiVO GUI and navigate to IPX configuration -> Configuration files. Then edit cid-superfecta.conf. Find the line that looks like the following and specify the sources you wish to use and the sequence in which they should be searched using the source numbers listed above to replace 0-1-2-3. Separate your entries with hyphens. Then SAVE the file.
same = n,AGI(nv-cid-superfecta.php,${XIVO_SRCNUM},0-1-2-3)
To use the commercial CNAM services of either OpenCNAM or BulkCNAM, you first must register with them and provide a credit card. You then will be provided credentials to use for your CNAM lookups. These need to be inserted at the top of /var/lib/asterisk/agi-bin/nv-cid-superfecta.php. Then SAVE the file.
Activating Traditional Asterisk Call Detail Recordings
If you want to preserve the numbers AND names of those that call your PBX, you’ll need to activate the traditional CDR reporting mechanisms in Asterisk®.
To activate SQLite3 logging of calls:
cd /etc/asterisk sed -i 's|no|yes|' cdr.conf echo "[master]" > cdr_sqlite3_custom.conf echo "table = cdr" >> cdr_sqlite3_custom.conf echo "columns => calldate, clid, dcontext, channel, dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags, accountcode, uniqueid, userfield" >> cdr_sqlite3_custom.conf echo "values => '${CDR(start)}','${CDR(clid)}','${CDR(dcontext)}','${CDR(channel)}', '${CDR(dstchannel)}','${CDR(lastapp)}','${CDR(lastdata)}','${CDR(duration)}', '${CDR(billsec)}','${CDR(disposition)}','${CDR(amaflags)}', '${CDR(accountcode)}','${CDR(uniqueid)}','${CDR(userfield)}'" >> cdr_sqlite3_custom.conf chown asterisk:www-data cdr_sqlite3_custom.conf chmod 660 cdr_sqlite3_custom.conf sed -i 's|noload => app_cdr.so|;noload => app_cdr.so|' modules.conf sed -i 's|noload => cdr_sqlite3_custom.so|;noload => cdr_sqlite3_custom.so|' modules.conf sed -i 's|noload => func_cdr.so|;noload => func_cdr.so.so|' modules.conf touch /var/log/asterisk/master.db chown asterisk:asterisk /var/log/asterisk/master.db chmod 640 /var/log/asterisk/master.db /etc/init.d/asterisk restart
To also activate CSV logging of calls:
cd /etc/asterisk echo "[csv]" >> cdr.conf echo "loguniqueid=yes" >> cdr.conf echo "loguserfield=yes" >> cdr.conf echo "accountlogs=yes" >> cdr.conf sed -i 's|noload => cdr_csv.so|;noload => cdr_csv.so|' modules.conf /etc/init.d/asterisk restart
To retrieve SQLite3 call log data, here are a few examples to get you started:
ALL: sqlite3 /var/log/asterisk/master.db "select * from cdr"
DATE: sqlite3 /var/log/asterisk/master.db "select * from cdr where calldate >= '2016-05-22'"
NPA: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE clid LIKE '%<843%'"
DEST: sqlite3 /var/log/asterisk/master.db "SELECT * from cdr WHERE dstchannel LIKE '%411%'"
FLDS: sqlite3 /var/log/asterisk/master.db "PRAGMA table_info(cdr)"
To retrieve the CDR log in CSV format suitable for spreadsheets, download:
/var/log/asterisk/cdr-csv/Master.csv
Adding Asterisk ULAW Sound Files to Your XiVO PBX
At least for us, the default sound files distributed with XiVO didn’t work. Here’s how to add the ulaw versions of all the files to your server:
cd /usr/share/asterisk/sounds/en wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-ulaw-current.tar.gz wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-ulaw-current.tar.gz tar zxvf asterisk-extra-sounds-en-ulaw-current.tar.gz tar zxvf asterisk-core-sounds-en-ulaw-current.tar.gz rm -f *.tar.gz chown asterisk:asterisk *.ulaw
Adding DISA Support to Your XiVO PBX
If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.
We use two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.
1. Download the DISA dialplan script into your /root folder where it can be edited:
cd /root wget http://incrediblepbx.com/disa-xivo.tar.gz tar zxvf disa-xivo.tar.gz rm -f disa-xivo.tar.gz nano -w disa-xivo.txt
2. When the editor opens the dialplan code, move the cursor down to the following line:
exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1) ; Good guy
3. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.
4. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:
exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)
5. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.
6. Now copy the dialplan code into your XiVO setup, remove any previous copies of the code, and restart Asterisk:
cd /root sed -i '\:// BEGIN DISA:,\:// END DISA:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf /etc/init.d/asterisk restart
7. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that we installed last week, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.
exten => 0,1(ivrsel-0),Dial(Local/3472@default)
8. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:
Goto(ivr-1,s,1)
Here’s how ours looks for the Nerd Vittles XiVO Demo IVR:
9. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.
SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.
Adding Vitelity to XiVO for Flawless VoIP Calling
We already have shown you several ways to take advantage of free VoIP calling in the U.S. and Canada as well as internationally. But, the old adage still holds true. You get what you pay for. And, if you’re using XiVO for your business or if you like a good night’s sleep without worrying about whether your spouse is going to stab you because of lousy phone connections, then splurge and spend a penny and a half a minute for outbound calls while getting unlimited incoming calls (4 at a time!) for only $3.99 a month. You’re worth it. The signup link for Vitelity is at the end of today’s article. Once you have your credentials, create a subaccount on the Vitelity site and then you’re ready to set up your Vitelity trunks with XiVO. We’ll use one trunk for incoming calls and a second trunk for outbound calls. The setup procedure for both trunks is already documented on the PIAF Forum. Make that your next stop!
PIONEERS: For Flawless VoIP, Choose Vitelity SIP Trunks for your XiVO PBX https://t.co/Kvovaj2f8F #asterisk #XiVO pic.twitter.com/XbUOP21GzV
— Ward Mundy (@NerdUno) May 16, 2016
Simultaneous Cellphone Ringing for Inbound Calls with XiVO
Speaking of incoming calls, wouldn’t it be nice if your cellphone also rang when XiVO calls arrived on your main extension. Then you don’t have to worry about missing a call just because you stepped out of the office.
If you took our earlier advice and purchased a RingPlus phone with free monthly service, then you’re already covered. Setting up the RingPlus SIP trunk last week covered all the bases. And, there’s more good news from RingPlus. Now you can buy a phone in their Classifieds section without previously owning a phone. So you can hit the ground running with a phone AND a free calling plan. For example, $149 currently buys a brand new Moto E with 3,000 4G/LTE and SIP minutes, 3,000 SMS messages, and 3,000 MB of LTE data every month. And the monthly cost: ZERO!
But, let’s assume you’re not the sharpest tool in the shed, and you still want your cellphone to ring when extension 701 rings on your PBX. Here’s how.
In the User setup for your extension:
1. Enter your cellphone number in the Mobile Phone Number field. Be sure it includes any necessary dial prefix so that it’s routed out through the correct trunk.
2. On the same screen, you’ll find a Preprocess subroutine field. Enter the following there: pre-mobility
3. SAVE your changes.
Keep in mind that outbound calls in XiVO are routed out using dialing prefixes. If you have set up a trunk with a provider that allows CallerID spoofing such as Vitelity, Anveo Direct, or VoIP.ms, then you can preserve the caller’s original CallerID number on the forwarded call to your mobile phone provided the dial string for your cellphone number matches the format you set up for the trunk you wish to use. For example, if Exten for Vitelity is 8NXXNXXXXXX, then you would enter the number for your cellphone with an 8 prefix: 89991234567.
Munin Makes XiVO Shine
If you look under the Services tab and choose Graphics, the World of Munin will suddenly appear. There are literally dozens of gorgeous charts to tell you anything and everything you’d ever want to know about your server’s performance. Enjoy!
Endpoint Management on Steroids… and It’s FREE
If you’ve longed for an endpoint manager that would automatically configure your phones, the wait is over. XiVO supports literally dozens of phones out of the box. And the setup is integrated into the setup procedure for the users and devices. To get started, choose the Configuration tab and click Plugins. Next click on the + icon to load the default endpoint config files. We couldn’t do justice to this topic in a blog. That’s what tutorials are for. And XiVO has a 700+ page reference guide that will tell you everything you ever wanted to know about endpoint management.
Adding NeoRouter VPN to XiVO
We’ll finish up for this week by showing you how easy it is to add the NeoRouter Client to XiVO. In less than five minutes, you’ll be able to use XiVO’s NeoRouter private IP address to access your server securely from anywhere in the world. Start by reading our last introduction to NeoRouter. If you do not already have a NeoRouter Server, follow this tutorial to set one up before you begin.
If you’re running XiVO on a 64-bit platform, issue the following commands to install the free NeoRouter client:
cd /root wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-amd64.deb dpkg -i nrclient-2.3.1.4360-free-ubuntu-amd64.deb
If you’re running XiVO on a 32-bit platform, do this instead:
cd /root wget http://download.neorouter.com/Downloads/NRFree/Update_2.3.1.4360/Linux/Ubuntu/nrclient-2.3.1.4360-free-ubuntu-i386.deb dpkg -i nrclient-2.3.1.4360-free-ubuntu-i386.deb
Unless you want your server identified in NeoRouter as localhost, we recommend changing your hostname and rebooting your server at this juncture. Just edit /etc/hostname and give it a name, e.g. xivo. Then reboot.
Now log back into your server as root and then log into your NeoRouter client. This will assign a private IP address to your XiVO server. The nrtap entry running ifconfig will tell you what that address actually is.
nrclientcmd ifconfig
Taking Nerd Vittles’ XiVO IVR for a Test Drive
There’s a Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles Demo IVR:
Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)
Published: Monday, May 23, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
The XiVO Adventure Continues: Adding Incredible PBX Goodies to Your Sandbox
We began our XiVO adventure last week by introducing a terrific new communications platform for both businesses and hobbyists. This week we begin the task of incorporating the Incredible PBX Goody Bag into an already amazing PBX, and we’ll cover about a dozen new topics. We’ll also address a few XiVO basics such as where to find and how to use the backups that XiVO makes every morning while many of us are still sleeping. Since a new XiVO release is imminent, we also want to show you how easy it is to upgrade your server. Before we get to the good stuff, we want to take a moment and document a fourth platform for XiVO that will appeal to many large organizations and perhaps some of our pioneers. It’s our platform of choice for development of new applications.
UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article superseded by the new release.
Installing XiVO as a VMware Virtual Machine
If your organization runs VMware, you may not need to worry about finding your own platform for XiVO. You can get your IT guys to build you a XiVO VM using XiVO’s Debian-based ISO. Then again, you might have followed our tutorial and chosen to run your own VMware ESXi server. In either case, a quick refresher on getting XiVO installed may be helpful. Begin by downloading XiVO to your Windows desktop. Then log into VMware vSphere Client on your Windows machine to access ESXi.
First, you’ll want to upload the XiVO ISO as a VMware guest operating system so that it can be used to create virtual machines at any time. From your inventory, click on the Configuration tab. Then click Storage under the Hardware listing. When your Datastore appears, right-click on datastore1 and choose Browse Datastore. Finally, click the Upload Files to this Datastore icon in your Datastore Browser and choose Upload File option. Choose the XiVO ISO from the Upload Items menu to upload it into your Datastore.
Now we’re ready to create a Virtual Machine. Right-click on the IP address of your VMware server and choose New Virtual Machine. Leave the Typical Configuration option selected and click Next. Give the virtual machine a name and click Next. Select the Destination Storage device and click Next. For the Operating System, choose Linux and pick Debian 8 (64-bit) then Next. Choose the NIC to use for the VM and click Next. Choose your Virtual Disk Size and Thin Provision option then Next. Check the box to Edit Virtual Machine Settings Before Completion and click Continue. Click the Options tab in Virtual Machine Properties and click Boot Options. Check the Force BIOS Setup option on next boot. Click Finish.
Starting your virtual machine the first time is not exactly intuitive so follow these steps carefully and in order. Keep in mind that, on the initial bootup of your virtual machine, what we want to do is run the XiVO ISO installer just as if we had booted a standalone machine using a CD on which we had burned the XiVO ISO. To begin the boot process correctly, first highlight your new VM by clicking on it and then choose Power on Virtual Machine. Next, click on the CD/DVD icon in the toolbar, choose CD/DVD Drive 1, choose connect to ISO image on Datastore. Double-click on datastore1 and then double-click on the XiVO ISO we uploaded previously. Now click on Launch Virtual Machine Console icon in the toolbar. When the BIOS setup utility appears, click in the window and use the Right Arrow key to move to the Boot tab. Move the CD-ROM option to the top of the list by highlighting it and pressing the + key to move it up. Press F10 to Save and Exit from the BIOS Setup and boot into your XiVO ISO. Click Install option to begin the regular XiVO installation procedure. When you finish the install, log into your server as root and obtain your IP address: ifconfig. You then can exit from the Console window by pressing Ctrl-Alt and use a browser to complete the install by pointing to the IP address of your virtual machine. Don’t forget that root SSH access is disabled by default. Our original tutorial will show you how to fix it AND install the Travelin’ Man 3 firewall whitelist to protect your server.
Adding a RingPlus SIP Trunk for Unified Communications with Sprint
Last week we began the XiVO adventure by turning on free Google Voice calling in the U.S. and Canada. Today we want to integrate smartphones into the mix by providing an incredibly simple and dirt cheap way to expand your XiVO communications platform while transparently meshing it with a RingPlus smartphone and the Sprint cellular network. When we’re finished, calls to your smartphone will also ring on one or more XiVO extensions. And designated users of your XiVO PBX will be able to place free calls to U.S. destinations using a SIP trunk tied directly to your RingPlus cellular account. These calls won’t be cellular. They’ll be pure VoIP calls using Sprint’s Internet backbone so listen for that pin to drop. If you have a (free) unlimited calling plan with RingPlus, then you’ll inherit a (free) unlimited calling plan for your XiVO PBX. Stated another way, whatever calling minutes you have with RingPlus can be shared on your XiVO PBX as inbound and outbound VoIP calls. The silver lining is that voicemails left on RingPlus get transcribed and delivered to your email address in seconds. So you get the best of both worlds. That’s what Unified Communications is all about!
Don’t worry if you’re late to the party and not yet a RingPlus user. They announce new deals every week so just check every few days until you find a plan that meets your needs. You won’t have to wait long. Here’s a list of all the previously announced PROMOS to give you a good handle on the scope of the RingPlus offerings. Deals don’t last but a couple hours or days so check often or sign up for RingPlus Alerts on SlickDeals and you’ll be the first to know! There’s a terrific deal tonight only from 8 p.m. until midnight.
Tonite @ 8 EDT: 3,750 Cellular/SIP monthly minutes free with $27 deposit https://t.co/fC4k9mhf5u #asterisk #RingPlus pic.twitter.com/huREsZVXC9
— Ward Mundy (@NerdUno) May 12, 2016
We’ve already documented the XiVO setup procedure on the PIAF Forum so hop over there to see how easy this is. Keep in mind that XiVO differs a bit from FreePBX® in the way Outbound Calls are managed. In FreePBX, you prioritized the routes by arranging them in a hierarchical list. In XiVO, you use unique dial strings, e.g. NXXNXXXXXX, for every Outbound Route. So, if you’re adding RingPlus to an existing XiVO server that already is using the NXXNXXXXXX dial string, then you’d need to use a different dial string to route calls out through the RingPlus trunk, e.g. 77NXXNXXXXXX with Stripnum=2. That tells XiVO that your users will dial calls to be handled by RingPlus with a prefix of 77 (RP), and then we want XiVO to strip off the first two digits before passing the call to the RingPlus SIP trunk for processing.
If you’re new to RingPlus, start with the original Nerd Vittles article for some background and then follow the RingPlus threads on the PIAF Forum and DSL Reports for the latest tips and tricks.
Adding a FreeVoipDeal (Betamax) SIP Trunk for Free International Calling
Before deploying a SIP trunk from one of the Betamax companies, read our latest article about Betamax for tips and tricks and land mines to watch out for. Then click the link below when you’re ready to deploy FreeVoipDeal as a trunk on your XiVO PBX:
PIONEERS: FreeVoIPDeal and Betamax Trunk Setups for XiVO PBX https://t.co/j4Q3vqSfmQ #asterisk #FreeCalling #voip pic.twitter.com/0axAv7Nmqn
— Ward Mundy (@NerdUno) May 13, 2016
Everything You Need to Know About XiVO Backups
Another feature of XiVO that separates the men from the boys is its documentation. In the case of backups, you’ll find everything you need to know here. All backups are stored on your XiVO server’s local drive in /var/backups/xivo. Be sure you have ample storage space available and, if you’re smart, you’ll copy both data.tgz and db.tgz from the local drive to a safe remote location periodically just in case disaster strikes. The documentation shows you how to quickly restore a backup should that ever become necessary.
Upgrading XiVO to the Latest Release
The XiVO development cycle is nothing short of miraculous. A new version is released every three weeks! The average time to close a bug has dropped from 315 days in 2009 to 28 days in 2012! You’ll probably want to keep your system current. 🙂
Upgrading XiVO is even easier than restoring a backup. Upgrade documentation is available here. Because we’ve added the Travelin’ Man 3 firewall, we recommend stopping IPtables during an upgrade and then restarting it when you’re finished. Your phone system is disabled during the upgrade. When upgrading XiVO, remember to also upgrade all associated XiVO Clients. Be sure to verify that things are back to normal once the upgrade procedure is completed: xivo-service status.
The commands to upgrade your XiVO PBX are as follows:
/etc/init.d/netfilter-persistent stop xivo-upgrade iptables-restart
Update: There’s a great tip from one of the XiVO developers on a better way to do this. See the first comment below.
Prerequisites for Today’s XiVO Adventure
If you’re just getting started with XiVO, DON’T START HERE. Read our first article. Be sure you have completed the following 8 steps before proceeding:
- Set Up Root SSH Access to Your XiVO PBX
- Set Up the Travelin’ Man 3 IPtables Firewall Using an SSH/Putty Connection
- Complete the XiVO Setup Using a Web Browser
- Create At Least One User with a 701 Extension
- Create At Least One SIP Trunk to Use for Outbound Calls
- Configure Outbound Call Settings for Your Trunk Using NXXNXXXXXX
- Configure an Inbound Route for Trunk Pointing to Your User Account
- If Behind NAT Firewall, Set externip and local network in General Settings -> SIP Protocol -> Network
Creating a MeetMe Conference Room for XiVO
There are just two steps to setting up a conference room. First, you need to add the extensions you will use for your conferences in the Default context. Then you add the Conference Room under IPBX Settings. Let’s set up a conference room extension 2663 (C-O-N-F). In your Default context, click on the Conference Rooms tab and enter an extension range of 2663-2664 and click Save. Then, in the Conference Rooms tab, click the + icon to add a new CONF conference room at extension 2663 in the Default context. You can experiment with the other settings when you have some spare time. The entries are pretty much self-explanatory. Click Save to activate your conference room. You won’t have music on hold for the first participant just yet. We’ll do that next.
Adding Music on Hold to XiVO
By default, XiVO doesn’t come with any music on hold. Fortunately, Digium has negotiated a music on hold license that you can use to add it to your PBX at no cost. While logged into your XiVO PBX as root, issue the following commands:
cd / wget http://incrediblepbx.com/moh-xivo.tar.gz tar zxvf moh-xivo.tar.gz /etc/init.d/asterisk restart
Asterisk Application Development with XiVO
For those coming from the FreePBX world, here’s a quick introduction to Asterisk application development on the XiVO platform. First and foremost, there are more similarities than differences. In the FreePBX environment, custom dialplan code was stored in /etc/asterisk/extensions_custom.conf. For custom extensions that you wanted to add, that code had to appear in the [from-internal-custom] context. For custom dialplan contexts, those appeared immediately below the last entry in the [from-internal-custom] context. If your custom code appeared anywhere else, there was always the risk that it might be overwritten with your next FreePBX reload.
The XiVO design is quite different. As we noted last week, it is not an Asterisk code generator at all, unlike FreePBX. Instead, it has a realtime interface to Asterisk using its PostGreSQL database engine. Updates are nearly instantaneous without reloading Asterisk modules from disk.
The other advantage is you won’t have to worry about XiVO stepping on your custom code as long as you leave PostGreSQL alone. HINT! The good news is there still are hooks to add your own custom dialplan extensions and code as well as PHP/AGI scripts. And it’s easy. In XiVO, custom extensions are stored in xivo-extrafeatures.conf which you’ll find in the /etc/asterisk/extensions_extra.d directory. Don’t edit files in /etc/asterisk/extensions_extra.d from the Linux command prompt! Instead, use the editor built into the XiVO GUI by selecting Configuration Files under IPBX configuration. This will automatically assure that realtime updates are posted correctly. To add additional contexts to your dialplan, create separate files for each context and store them in this same directory. Again, the easy way to make certain that Asterisk is updated automatically when you add new code snippets is to create and edit them within the XiVO GUI. These files all will appear under IPBX Configuration -> Configuration Files as well.
In order to better mimic the FreePBX way of doing things so that your PHP/AGI scripts work in either environment, we recommend issuing the following symlink while logged into XiVO. We’ll do it as part of the SQLite3 install below.
ln -s /var/lib/asterisk/agi-bin /usr/share/asterisk/agi-bin
Once you’ve established the symlink, PHP/AGI scripts can be migrated from FreePBX to XiVO directly using the same directory structure for storage: /var/lib/asterisk/agi-bin. As with FreePBX, all files in this directory should be owned by asterisk with 775 permissions:
chown asterisk:asterisk /var/lib/asterisk/agi-bin/* chmod 775 /var/lib/asterisk/agi-bin/*
There are many other powerful features in XiVO that weren’t available at all in FreePBX. We’ll cover some of them in coming months. In the meantime, this brief overview of the dialplan environment should be sufficient to let you start building.
Installing SQLite3 to Support Incredible PBX Applications
There’s one other difference between XiVO and FreePBX that we’ve already touched upon. But it bears repeating here. XiVO doesn’t use MySQL or MariaDB for its database management tasks. Instead, the XiVO development team chose PostGreSQL which is equally powerful, but different. For the Incredible PBX application suite, we’ve chosen to rewrite the ones that depend upon MySQL so that they can run under SQLite3 which is considerably less processor intensive than running both PostGreSQL and MySQL 24/7. We also didn’t want to interfere with the PostGreSQL setup of XiVO since it is an integral component of the product and will get upgraded automatically as part of the regular XiVO upgrade cycle.
Here’s how to put the SQLite3 and corresponding ODBC components in place on your new server. While logged into your server as root, simply issue the following commands:
cd / wget http://incrediblepbx.com/sqlite3-xivo.tar.gz tar zxvf sqlite3-xivo* rm -f sqlite3-xivo.tar.gz cd /root ./sqlite3-xivo.sh
Running a couple SQLite3 queries using the ZIPCODES and ASTERIDEX databases will give you a feel for the performance you can expect from SQLite3. The queries might look like this:
sqlite3 /var/lib/asterisk/agi-bin/zipcodes.sqlite "select zip,city,state from zipcodes where zip=29401;" sqlite3 /var/lib/asterisk/agi-bin/asteridex.sqlite 'select name,out from user1 where name LIKE "%Airlines%";'
And here are the results of the two queries:
29401|CHARLESTON|SC -------------------------------- American Airlines|8004337300 Continental Airlines|8005250280 Delta AirLines|8002211212 Frontier Airlines|8004321359 Iberia AirLines|8007724642 Midway Airlines|8004464392 Northwest Airlines|8002252525 Southwest Airlines|8004359792 Ted Airlines|8002255833 United Airlines|8002416522 WestJet Airlines|8005385696 Yemen Airlines|8009368300
We’ve included a bonus script in /root that will let you convert existing MySQL databases to SQLite3. For example, if you’re currently using AsteriDex on another Incredible PBX platform, it only takes a couple seconds to convert your MySQL database to SQLite3. The syntax to run the script should look like this:
./mysql2sqlite3.sh -u root -ppassw0rd yourdatabase | sqlite3 yourdatabase.sqlite
You obviously cannot run the script on your XiVO server because your MySQL databases and MySQL itself are missing. So move the script to the server on which your MySQL databases are stored and run it there using the above syntax. Then copy the asteridex.sqlite file to your XiVO server and save it in /var/lib/asterisk/agi-bin.
Installing and Activating the Festival TTS Engine with Asterisk
We’ve got a couple more building blocks to put in place to support Incredible PBX applications. Then we’ll be ready to kick the tires with a few applications to get you started. In coming weeks, we’ll finish up the conversion of the remaining apps, and then we’ll publish an Incredible PBX installer for XiVO with all the pieces. But why wait? Finish up installing the remaining pieces today, and you’ll have something to play with. And, as we said, it will also provide you with simple scripts so you can actually see how Incredible PBX is put together.
Many of the Incredible PBX applications rely upon text-to-speech and/or voice recognition (speech-to-text) to work their magic. Neither comes installed with XiVO by default, but Asterisk was properly configured to support Festival so let’s work with that. Festival is the Big Brother of FLITE and includes some additional voices of fairly good quality. The XiVO Demo IVR will give you an idea of the TTS voice quality you can expect:
To get Festival installed and activated for use with Asterisk, issue these commands:
cd / wget http://incrediblepbx.com/festival-xivo.tar.gz tar zxvf festival-xivo.tar.gz cd /root ./festival-xivo.sh
Installing Dial Plan Code for Sample Incredible PBX Applications
Now we’re ready to put today’s Dial Plan Code and IVR in place and load the PHP/AGI components necessary to make the sample applications work. Here’s how:
cd / wget http://incrediblepbx.com/ivr-xivo.tar.gz tar zxvf ivr-xivo.tar.gz chown asterisk:www-data /etc/asterisk/extensions_extra.d chmod 775 /etc/asterisk/extensions_extra.d chmod g+s /etc/asterisk/extensions_extra.d /etc/init.d/asterisk restart
Installing and Activating Voice Recognition for XiVO
Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your XiVO PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Don’t forget to add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.
Taking XiVO on a Test Drive with the Incredible PBX Apps
Now set up a softphone using the IP address of your XiVO server and the Line credentials for Extension 701. When you obtain your credentials, double-check to make sure all of the fields for the Line are filled in correctly as shown below:
Once your softphone is registered, you can try out some of the sample applications:
- 4871 (IVR1) – Allison’s Demo IVR
- 411 (Voice Dialing) – Call by Name (try "Delta Airlines")
- 2663 (CONF) – MeetMe Conference with Music on Hold
- 951 – Yahoo! News Headlines (TTS)
- 947 (ZIP) – NWS Weather by ZIP Code
- 53669 (LENNY) – The Telemarketer’s Worst Nightmare
You can review the Dialplan code in the GUI by choosing Configuration Files and clicking xivo-extrafeatures.conf. The sample IVR code is in ivr-1.conf.
Taking Nerd Vittles’ XiVO IVR for a Test Drive
There’s also a new Demo IVR running at www.pacificnx.com on their XenServer virtualization platform. Scott McCarthy, a leading outside XiVO developer and a principal at PacificNX, tells us they soon will have a $20 a month platform specifically tailored to XiVO. And that’s what you’ll be hearing when you call the Nerd Vittles IVR: 1-843-606-0555. Setup at PacificNX took less than a minute. Enjoy!
Published: Thursday, May 12, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business
We promised you that 2016 was going to be a year filled with surprises, and today we’re pleased to introduce another open source, pure GPL3 solution for any business. Whether your requirements are a call center or a versatile phone system for hundreds of employees, XiVO™ offers a compelling unified communications solution that checks all the boxes. Unlike some products that function merely as a code generator for Asterisk®, XiVO is in a league of its own. XiVO is actually an integral component of the Asterisk application itself. It manages your telephony server in realtime using its versatile PostGreSQL database platform. Did we mention it’s also a great playground for hobbyists and SOHO VoIP enthusiasts? Let’s get started.
UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article superseded by the new release.
There’s no way to do justice to a product like XiVO in a single article. So our plan is to introduce XiVO today and get your platform up and running where you can make and receive free calls throughout the United States and Canada. Then you can add Incredible PBX components and additional SIP providers as we continue to build them out. Just follow along with our Incredible PBX development for XiVO on the PIAF Forum, and you’ll get a first-hand look at how sausage is made. We already have text-to-speech applications for news and weather up and running. You can take them for a test drive by calling the XiVO demo:
And, of course, we’ve integrated the Travelin’ Man 3 IPtables firewall to provide rock-solid security for XiVO, and we’ll cover that today as well. As part of this development process, you’ll discover how easy it is to build Asterisk applications for XiVO on your own. And hopefully you’ll share some of your creations with the rest of us. That’s what open source development is all about.
Choosing an Experimental Platform for XiVO
We’re just getting started with XiVO development so, like us, we’re assuming you’ll want to kick the tires a bit before jumping into a new VoIP solution for the long haul. That means you first must choose a platform on which to install XiVO. We have several recommendations for you. If you have a robust desktop machine with lots of RAM and processing power, then installing XiVO under VirtualBox may be the way to go. We actually use an iMac with 16GB of RAM, and it provides plenty of horsepower to run VirtualBox and XiVO. With VirtualBox, we’ll start by downloading the XiVO ISO.
We didn’t mention that XiVO has been under development for over 10 years and is supported by the original developers with financial support from Avencall. Because of its Canadian roots, it seems only fitting that many may wish to consider CloudAtCost in Canada as an appropriate site to host your experimental XiVO server. A one-time payment of $10.50 still buys you a sandbox in the cloud for life with coupon code TAKE70, and XiVO installs on the CloudAtCost platform without a hiccup. For a CloudAtCost implementation, we’ll start by creating a Debian 8 server.1 And then we’ll download and run the XiVO installation script to build our XiVO server. Finally, we’ll walk you through setting up XiVO on a $5/month Digital Ocean Droplet which provides state-of-the-art performance at rock-bottom Cloud pricing. So begin by choosing your hardware platform from the three options below:
1. Installing XiVO as a VirtualBox Virtual Machine
For standalone implementations including VirtualBox, we’ll begin by downloading the 64-bit XiVO Server ISO to your desktop. Next, create a VirtualBox 64-bit Debian VM platform with 1024 MB RAM and at least a 10GB virtual drive. In System Settings, enable I/O APIC and disable the other options. Select a Sound Card to match your machine and configure Network Adapter 1 as a Bridged Network Device. In the Storage Settings (shown below) for your (1) Empty IDE Controller, (2) select the downloaded XiVO ISO as your installation media. Start the VM and proceed through the initial install.
Click Install, choose your language, pick your time zone, choose your keyboard map, create a very secure root password, and choose a Debian mirror that’s close to your server. Choose /dev/sda as your bootloader assuming that’s the disk drive configured by VirtualBox. In less than 10 minutes, the install will complete and your VM will reboot. Log into your server as root and obtain your IP address: ifconfig. You’ll need it for the web configuration step that comes next.
2. Installing XiVO as a CloudAtCost Cloud-Based Server
You can’t use an ISO as the installation media at CloudAtCost so we have to start by building a 64-bit Debian 8 virtual machine with at least 512 MB RAM and a 10GB virtual drive. No need to choose a larger drive at the moment since there’s a bug in CloudAtCost’s installer for Debian 8. See the footnote for details. Once your virtual machine is built, log in as root and issue the following commands to kick off the XiVO install:
apt-get -y remove apache2* apt-get update apt-get -y upgrade reboot # log back in as root and... wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh bash xivo_install_current.sh
3. Installing XiVO as a Digital Ocean Droplet
As with CloudAtCost, you’ll need to begin your XiVO adventure at Digital Ocean by first signing up for an account. With our referral code, you’ll get a $10 credit (and so will Nerd Vittles). That’s good for two full months of service to kick the tires of XiVO without ever spending a dime. Once your account is set up, create a $5/month Debian 8 (64-bit) Droplet. When you receive the email with your droplet credentials, log into your new server as root using SSH/Putty and issue the following commands to get Debian 8 squared away:
apt-get update apt-get upgrade -y dd if=/dev/zero of=/swapfile bs=1024 count=1024k chown root:root /swapfile chmod 0600 /swapfile mkswap /swapfile swapon /swapfile echo "/swapfile swap swap defaults 0 0" >> /etc/fstab sysctl vm.swappiness=10 echo vm.swappiness=10 >> /etc/sysctl.conf free reboot
After the reboot, log into your server again with your new root password and kick off the XiVO install:
wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh bash xivo_install_current.sh
Enabling SSH Root Access with XiVO
If you installed XiVO using the XiVO ISO, then root logins via SSH are disabled by default. Only enable it if you plan to also implement the firewall in the next step! To enable root logins via SSH, log into the server console as root and edit the SSH config file: nano -w /etc/ssh/sshd_config. Find the line in the Authentication section that begins with PermitRootLogin and change it to: PermitRootLogin yes. Save your change (Ctrl-X, y, ENTER) and then restart SSH: /etc/init.d/ssh restart.
Setting Up a Firewall to Protect XiVO
We don’t build PBXs without a rock-solid firewall, but it’s your phone bill so the choice is all yours. The Travelin’ Man 3 implementation of the Linux IPtables firewall provides a safe computing platform using a WhiteList to only allow access by trusted users and providers. You can add additional users to the whitelist as desired using add-ip and add-fqdn in the /root folder. Restart your firewall using only this command: iptables-restart. If you’ll be using FQDNs in your WhiteList, then add the ipchecker script to your cron jobs. Then review Step #5 in the TM3 tutorial.
echo "*/10 5-22 * * * root /root/ipchecker > /dev/null 2>&1" >> /etc/crontab
It’s imperative that you set this up from a client workstation that’s running SSH or Putty. Otherwise, you may inadvertently lock yourself out from your own server. While logged into your server via SSH as root, issue the following commands:
cd /root wget http://incrediblepbx.com/firewall-xivo.tar.gz tar zxvf firewall-xivo.tar.gz rm -f firewall-xivo.tar.gz ./tm3-xivo.sh
Configuring XiVO with a Web Browser
Once the basic install is completed, you use a web browser to actually configure and manage your XiVO server. To get things started, point your browser to the IP address of your XiVO server. Choose your Language. Accept the GPL3 license agreement. Then fill in the blanks to create a Hostname for your server (XiVO), a domain name (some domain that you own or one chosen from your favorite dynamic DNS provider), a very secure Web interface password (choose as if your phone bill depends upon it). The network interface and DNS server entries should already be correct. Click Next.
On the second configuration screen, choose an Entity (department/organization name or IncrediblePBX will suffice). Then set up the Contexts to manage calls on your PBX:
- Internal Calls Context: manages extension numbers that can be reached internally
- Incalls Context: manages calls coming from outside of your system
- Outcalls Context: manages calls going from your system to the outside
Here’s what we’ll be using by way of example:
Finally, validate your entries to complete the configuration. Now log into your XiVO server as root using your newly created web password. You should get a status screen that looks something like this. If you had any doubts about the quality of the XiVO product, this should put your mind at ease. 🙂
Logging Into the XiVO Web Interface
To make changes in your XiVO setup, you’ll need to log into the web interface at the IP address of your XiVO PBX. Login with root as the username together with the Web Interface Password you set up above. You can change this password at any time under the Configuration tab by clicking on Users and editing your existing settings.
Creating Users and Lines with XiVO
For those migrating from the FreePBX® world, you’re probably most familiar with the procedure for creating extensions. More advanced administrators may have switched to device and user mode where users and devices are created separately. Phone numbers or extensions were associated with users while phone instruments were associated with devices. In the World of XiVO, we’ll start with the simplest configuration, and you can move on from there when you’re ready. In our scenario today, we’ll create a couple of users. Each user has a Name, Language, Time Zone, and other optional characteristics such as a Mobile Phone Number which can ring simultaneously whenever a user receives a call to his or her local XiVO phone number. By adding a Line (aka Phone Number) for the user as the user account is created, XiVO will automatically generate a separate Line with username and password credentials. This Line will be associated with the User during the initial user setup procedure, and this Line then can be registered to a SIP phone, softphone, or XiVO client (which we will cover separately down the road). In the example below, we’re using Nerd Uno’s extension 701 (associated with line 3jz8tsr0) to call Nerd Dos’ extension 702 (associated with line 8fmne2x4).
XiVO has an excellent tutorial that covers creating Users with a SIP Line. So jump there and add a couple of Users following the steps in the tutorial. When you’re finished, you’ll have two Users and two associated Lines with credentials to set up SIP phones. Since you’re just getting your feet wet and will probably make some mistakes, it’s probably a good idea to turn off Fail2Ban while you’re experimenting. Otherwise, you may accidentally lock yourself out of your server (ask us how we know) and think it’s a problem with XiVO. Here’s how:
/etc/init.d/fail2ban stop
To set up your SIP phones, you’ll need the credentials for each of the two lines. Under the Lines tab, click on the Pencil icon to reveal the Username and Password. Fill in the missing pieces as shown below and make certain that your NAT entry is set to Yes.
With those credentials in hand, go ahead and configure a couple of SIP phones and make certain you can call between them with audio in both directions before proceeding. For those with a Mac, Telephone is perfect for experimentation because you can set up multiple softphones and place calls between them.
IMPORTANT: If your server is sitting behind a NAT-based firewall, you must set the external and local network IP addresses for XiVO in General Settings -> SIP Protocol. You’ll find the fields in the Network tab.
Configuring a SIP Trunk for Google Voice with XiVO
Now that you have internal calls working, let’s turn our attention to connecting your PBX to the rest of the world. We obviously can’t cover the setup for every SIP provider, but we can provide a good example that will get our U.S. friends free calling in the U.S. and Canada. We’ve chosen the Simonics SIP Gateway to Google Voice because a one-time payment of $5.99 gets you a traditional SIP trunk to interface with any existing Google Voice number. If you don’t have a Google Voice number, sign up here. In your Google Voice Settings, make sure Forward Calls to Google Chat is enabled and disable Call Screening in the Calls tab. Then, with your Google credentials and Google Voice number in hand, visit the Simonics web site to sign up for service. Sign in with your Google credentials and complete the registration process. Once you have your Simonics account name and password, log into your XiVO web portal.
With credentials in hand, on the XiVO side, start by choosing the SIP Protocol tab under Trunk Management. There are actually three tabs to configure for the SIP trunk. Begin in the General tab and make it look like this using your credentials. NOTE: The complete FQDN for the Simonics gateway should be gvgw.simonics.com:
Next, click on the Register tab and reenter your credentials. Leave the empty fields exactly as shown. Be sure the Register box is checked.
Next, in the Signaling tab, change the Monitoring option to Yes and then click Save. Monitoring is the XiVO equivalent of the SIP Qualify option.
We also need to make one minor adjustment in the SIP Protocol Defaults in the General Settings. Just Save your settings after checking Match users with ‘username’ field.
Next, we need to tell XiVO how to process Incoming and Outgoing Calls using the Google Voice SIP trunk. Under the Call Management section, let’s begin with the Incoming Calls setup by creating a new Incoming Calls DID for your 11-digit Google Voice number. To keep things simple, we’ll route the incoming calls to the User mapped to extension 701:
For Outgoing Calls, we need to route calls with a specific dial string out the Simonics SIP trunk using the to-extern context. By way of example, we’ve set this up using a dialing prefix of 48 (GV) and a 10-digit number. We’re letting XiVO supply the missing 1 country code required by Google Voice, and we’ll let XiVO strip off the 48 prefix in processing the outbound calls. If this is your only outgoing trunk, you may prefer not to use a dial prefix at all. In that case, change the dial string to a 10-digit number (NXXNXXXXXX) and set Stripnum to 0.
Well, that’s enough for today. There’s complete XiVO PDF Documentation available here. We’ll have lots more to say about XiVO in coming weeks. Come join the party!
Published: Thursday, May 5, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
- There’s a glitch in the CloudAtCost builds for Debian8. Regardless of how much disk storage you allocate, CloudAtCost will only use 10GB. Moral: Don’t waste your resources by allocating more than 10GB of disk space. This is an experimental platform, and 10GB will suffice. If you really need more space, this thread on the PIAF Forum will walk you through expanding the storage allocation beyond the 10GB threshold. [↩]
Chasing the Asterisk Rainbow: An Incredible PBX for Everyman… and Woman
The one thing we’ve learned over the past decade is that everyone’s telecommunications needs are different. We began the Incredible PBX™ project by offering a script for PBX in a Flash™ that enhanced its functionality by adding three dozen applications for Asterisk® plus a rock-solid, preconfigured firewall. We expanded on that initial design by offering installers to support both Elastix® 4.0 and even the FreePBX® Distro and AsteriskNOW. But many asked for something more. Some wanted a turnkey install on inexpensive hardware so we now have a Raspberry Pi® 3 image that configures itself in under a minute. Others wanted and we provided an ISO image that could be loaded onto a USB thumb drive and used to install Incredible PBX on virtually any standalone hardware platform including the $200 Intel NUC. Then there were the enthusiasts that insisted upon virtual machines for virtually everything. So we built Incredible PBX images for Proxmox, and VirtualBox, and VMware. All of these builds could be set up about as quick as you can snap your fingers. Still others believed that all computing should be handled with Cloud-based servers. So we created a $10.50 solution for lifetime Cloud computing with CloudAtCost, a $5/month solution with Digital Ocean, and a $15/month rock-solid reliable PBX with RentPBX. Finally, there were operating system purists. Some preferred RedHat/CentOS/Scientific Linux while others swore by Debian or Ubuntu or Raspbian 8. So we added Incredible PBX builds for every conceivable operating system.
What remained consistent through all of these Incredible PBX iterations was our absolute commitment to providing a secure computing environment out of the box with a feature set unmatched in both the open source and commercial PBX communities. But, of course, there is always someone that doesn’t want features and prefers a secure platform on which to build their own servers. We’ve even accommodated that request with the Lean, Mean Edition of Incredible PBX.
So, today while we work on taxes, you can take a leisurely stroll through all of the available Incredible PBX links above and choose a platform that best meets your needs. Our other uncompromising feature is price. Incredible PBX is and always will be free with NoGotchas. If you’re still confused about choosing a favorite build, try our Decision Tree below. It’ll give you suggestions in less than a minute. Come join the party!
Originally published: Monday, April 11, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
The Sensational Raspberry Pi 3 Featuring Incredible PBX GUI with Raspbian 8 Jessie
[iframe-popup id="3″]
Hard to believe it’s been four years since the introduction of the original Raspberry Pi®. Over eight million RasPi’s have been shipped. To celebrate its fourth birthday, Eben Upton has done it again. Meet the sensational Raspberry Pi 3 sporting a 1.2GHz 64-bit quad-core ARM Cortex-A53 CPU with ten times the performance of the original Raspberry Pi. Of particular interest to the VoIP community will be the RasPi 3’s integrated 802.11n wireless LAN and Bluetooth 4.1 hardware. And, of course, the RasPi 3 retains its compatibility with the Raspberry Pi 1 and 2. Did we mention it’s still just $35? Because we like to celebrate birthdays, too, we’re pleased to introduce a brand new Incredible PBX™ image for the Raspberry Pi 2 and 3 featuring Raspbian 8 and the latest release of Asterisk® 13. Unlike previous builds, this one installs in under a minute. Yes, it’s still FREE and features pure open source GPL code. No Gotchas!
07/01/2019 NEWS FLASH: Just released Incredible PBX LITE for the Raspberry Pi 2, 3, and 4 featuring Raspbian 10 Buster. Tutorial here.
Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. Even with a Raspberry Pi 2, we have detected no performance degradation thanks to the latest Raspbian 8 OS and a virtually flawless Asterisk 13 platform. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful at the $10 price point.1
Raspberry Pi 3 Shopping List. Before you can install Incredible PBX, you’ll need a compatible Raspberry Pi 3 platform. Here’s the short list:
$35* Raspberry Pi 3 from MCM or Newark or Amazon $10 Power Adapter (2.5 amps minimum!) $9 32GB microSDHC Class 10 card £12.95 Rainbow Pibow case or $9.50 Official RasPi case
About That Asterisk. We write about Asterisk® regularly, but the asterisk we’re talking about is the one accompanying the $35* price tag for the Raspberry Pi 3. Yes, that’s the advertised price. And, no, if you want one this year, you’re not going to pay that. There are the marked up shipping prices, the bundled add-on’s that you don’t need or want, and the must-have accessories like a power adapter. We’re assuming you already own a USB keyboard and an HDMI-compatible monitor. If so, just plan on $100 and consider yourself lucky if you get all the pieces for less. Our order from Pimoroni in the U.K. with a case and 3-day shipping was £59.36 or $82.95 U.S. Our order from MCM for just the RasPi 3 with shipping was $46.99.
Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, GVSIP, SMS, and SRTP functionality. Voice Recognition and Text-to-Speech VoIP application support using FLITE, GoogleTTS, and PicoTTS. Free calling with Google Voice, Simonics SIP gateway, or RingPlus cellular service. And all of your Nerd Vittles favorites: Fax, AsteriDex, Click-to-Dial, News, Weather, Reminders, and Wakeup Calls. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Email Delivery of Voicemail, Voicemail Blasting, and more…
10-Layer Network Security Model. Most phone calls cost money. Unlike many of the other "free" VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, it isn’t free at all. Once you plug in that network cable, you’ve painted a bullseye on your checkbook.
No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in Apache, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. Ours is all open source GPL code so we would encourage everyone to get on board and do their part to make the Internet a safer place!
Do your part and do your homework. Comparison shop as if your phone bill matters! 😉 Incredible PBX provides:
- Preconfigured IPtables Linux Firewall
- Preconfigured Travelin’ Man 3 WhiteLists
- Randomized Port Knocker for Remote Access
- TM4 WhiteListing by Telephone (optional)
- Fail2Ban Log Monitoring for SSH, Apache, Asterisk
- Randomized Ultra-Secure Passwords
- Automatic Update Utility for Security & Bug Fixes
- Asterisk Manager Lockdown to localhost
- Apache htaccess Security for Vulnerable Web Apps
- Security Alerts via RSS Feeds in Kennonsoft and Incredible PBX GUIs
Installation Tutorial. Here’s everything need to know about installation and setup. "Automatic" means you just watch.
- Download and unzip Incredible PBX image from SourceForge (with or without GV OAuth support)
- Transfer Incredible PBX image to microSD card
- Boot Raspberry Pi from new microSD card
- Login to RasPi console as pi:raspberry to initialize your server (Automatic)
- Reboot after writing down your server IP address (Automatic)
- Login via SSH as root:password to randomize passwords & configure firewall (Automatic)
- Optionally, install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)
- Enjoy!
Configuring Trunks with Incredible PBX
Before you can actually make and receive calls, you’ll need to add one or more VoIP trunks with providers, create extensions for your phones, and add inbound and outbound routes that link your extensions to your trunks. Here’s how a PBX works. Phones connect to extensions. Extensions connect to outbound routes that direct calls to specific trunks, a.k.a. commercial providers that complete your outbound calls to any phone in the world. Coming the other way, incoming calls are directed to your phone number, otherwise known as a DID. DIDs are assigned by providers and you register your trunks using credentials handed out by these providers. Incoming calls are routed to your DIDs which use inbound routes telling the PBX how to direct the calls internally. A call could go to an extension to ring a phone, or it could go to a group of extensions known as a ring group to ring a group of phones. It could also go to a conference that joins multiple people into a single call. Finally, it could be routed to an IVR or AutoAttendant providing a list of options from which callers could choose by pressing various keys on their phone.
We’ve done most of the prep work for you with Incredible PBX. We’ve set up an Extension to which you can connect a SIP phone or softphone. We’ve set up an Inbound Route that, by default, sends all incoming calls to a Demo IVR. And we’ve built a dozen trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re good to go.
Unlike traditional telephone service, you need not and probably 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 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, all you need are your credentials for each provider and the domain name of their server. Log into Incredible PBX GUI Administration as admin using a browser. From the System Status menu, click Connectivity -> Trunks. Click on each provider you have chosen and fill in your credentials including the host entry. Be sure to uncheck the Disable Trunk checkbox! Fill in the appropriate information for the Register String. Save your settings by clicking Submit Changes. Then click the red Apply Config button.
Configuring a Softphone for Incredible PBX
We’re in the home stretch now. You can connect virtually any kind of telephone to your new 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 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Applications _> Extensions -> 701 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, 701 for your Username, and whatever Password you assigned to the extension when you installed Incredible PBX. Click OK to save your entries.
Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:
DEMO - Apps 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: Do NOT under any circumstances take Google’s bait to switch from Google Chat to Hangouts, or you will forever lose the ability to use Google Chat with Incredible PBX. 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 Incredible PBX 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 Screening – OFF
- Call Presentation – OFF
- Caller ID (In) – Display Caller’s Number
- Caller ID (Out) – Don’t Change Anything
- Do Not Disturb – OFF
- Call Options (Enable Recording) – OFF
- Global Spam Filtering – ON
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 the Incredible PBX GUI by choosing Connectivity -> Google Voice. How you enter your credentials depends upon whether you have chosen the Incredible PBX image with OAuth 2 support. For a complete Google Voice OAuth tutorial, follow steps 8-10 in this Nerd Vittles tutorial. Once you’ve entered your credentials, you MUST restart Asterisk from the command line, or Google Voice calls will fail.
If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
If you have difficulty finding the Google Chat option after setting up a new Google Voice account, follow this tutorial.
Another option is to use an inexpensive SIP Gateway to Google Voice. The Simonics trunk in the Incredible PBX GUI is preconfigured for this purpose. All you’ll need is your Google Voice credentials. Get started with this tutorial.
Adding Speech Recognition Support to Incredible PBX
To support many of our applications, Incredible PBX has included Google’s speech recognition service for years. These applications include Weather Reports by City (949), AsteriDex Voice Dialing by Name (411), and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for "personal and development use." If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!
To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Do NOT change the yourID portion of the command:
sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747
Now you’re ready to try out the speech recognition apps. Dial 949 and say the name of a city and state/province/country to get a current weather forecast from Yahoo. Dial 411 and say "American Airlines" to be connected to American.
To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. "What planes are overhead." Read the Nerd Vittles tutorial for additional examples and tips.
Enabling WiFi on the Raspberry Pi
With the Raspberry Pi 3, wi-fi hardware is included. With the Raspberry Pi 2, you’ll need to add an inexpensive wifi dongle. The next step is connecting to your WiFi router. Simply open /etc/wpa_supplicant/wpa_supplicant.conf with your favorite editor and insert the following code using the actual SSID name and password to access your local, password-protected WiFi router or any open WiFi network:
network={ ssid="YourSSID" psk="YourSSIDpassword" key_mgmt=WPA-PSK scan_ssid=1 priority=5 } network={ key_mgmt=NONE priority=1 }
Finally, stop and restart the wlan0 interface, count to 15, and check the status of your server to decipher the new IP address for your WiFi connection:
ifdown wlan0 ifup wlan0 pbxstatus
If you want to run your Raspberry Pi exclusively off the WiFi connection, simply unplug the network cable from your RasPi and reboot your server.
UPDATE: There still is a quirk with the wireless LAN driver on the Raspberry Pi 3. The problem has to do with the default power management of the wlan0 interface which results in it being powered off after very brief periods of inactivity. Special thanks to Matt Gemmell for this fix. Just cut-and-paste the lines below into a terminal window, and you’ll be good to go.
WARNING: Run pbxstatus first. If the top line shows Raspberry Pi 3, the following WiFi patch is already installed.
echo "options 8192cu rtw_power_mgnt=0 rtw_enusbss=0 rtw_ips_mode=1" > /etc/modprobe.d/8192cu.conf sed -i '/exit 0/d' /etc/rc.local echo "sleep 10" >> /etc/rc.local echo "iwconfig wlan0 power off" >> /etc/rc.local echo "exit 0" >> /etc/rc.local echo "[Unit]" > /etc/systemd/system/root-resume.service echo "Description=Turn off wlan power management" >> /etc/systemd/system/root-resume.service echo "After=suspend.target" >> /etc/systemd/system/root-resume.service echo "" >> /etc/systemd/system/root-resume.service echo "[Service]" >> /etc/systemd/system/root-resume.service echo "Type=simple" >> /etc/systemd/system/root-resume.service echo "ExecStartPre= /bin/sleep 10" >> /etc/systemd/system/root-resume.service echo "ExecStart= /sbin/iwconfig wlan0 power off" >> /etc/systemd/system/root-resume.service echo "" >> /etc/systemd/system/root-resume.service echo "[Install]" >> /etc/systemd/system/root-resume.service echo "WantedBy=suspend.target" >> /etc/systemd/system/root-resume.service systemctl enable root-resume reboot
After rebooting, if you issue the iwconfig wlan0 command, it should show: Power Management:off.
Update: Lessons Learned for Raspberry Pi 3 Road Warriors
As with all new devices, you learn some things as you go along. So we’re providing an update to our original article to offer a couple of additional tips and tricks for those that want to travel with a RasPi…
Alternative Power Sources. If you’re like us, you have a number of devices around the house or office that all require 5V power adapters of various amperages. The Raspberry Pi has traditionally been one of the most temperamental when it came to power adapters and, with the Raspberry Pi 3, the developers specifically mention a 2.5 amp minimum. If you travel and want to take devices such as the Raspberry Pi with you, the last thing you want to do is approach airport security with a bunch of wires hanging out of your carry-on bag. Well, there’s good news. The Anker device shown in the Amazon ad in the right column of Nerd Vittles can supply power to 6 devices including a Raspberry Pi 3. And we’ve given the RasPi a healthy workout with no adverse effects.
Deciphering the RasPi IP Address. As we mentioned, we travel a lot so obtaining a DHCP address for your RasPi in WiFi mode is not always the easiest thing to accomplish. If your smartphone supports tethering, that’s the easiest way to get connectivity on the road. A better way is to stick a WiFi HotSpot in your luggage and it, too, can be powered using the Anker device. See our recent article for WiFi HotSpot choices. Regardless of which option you choose, it will require some planning to use your RasPi sans monitor and keyboard. First, you need to preconfigure /etc/wpa_supplicant/wpa_supplicant.conf with the SSID of the device you’ll be using to hand out DHCP addresses. You’ll note from the discussion above that each entry in this file has a priority with higher numbers having higher priority. The way we typically do this is to assign our home network as the highest priority. Below that, we set up credentials for our MiFi Hotspot, then our smartphones, and finally open networks. So it looks like this:
- Home Network – 6
- MiFi HotSpot – 5
- Android phone – 4
- iPhone (AT&T) – 3
- Open Network – 1
Keep in mind that the Incredible PBX firewall probably will block you from accessing the RasPi from a computer on the public network. So you also must connect your computer to the same private WiFi network because private LAN addresses are whitelisted in the firewall by default.
Once you have connectivity for your RasPi and your laptop, the other wrinkle is figuring out the IP address of the Raspberry Pi. Our recommended approach goes like this. First, configure SendMail on the RasPi to use a Gmail account that you own as an SMTP smarthost to send emails. That should work almost anywhere you go. Second, modify /etc/rc.local to automatically send you an email with the IP address and SSID of your wireless network whenever the RasPi boots. Again, this takes some advance planning because you need to set all of this up and test it before you go on the road.
Here are the steps to modify SendMail to use an existing Gmail account as a SmartHost. Log into your RasPi as root and issue the following commands:
cd /etc/mail hostname -f > genericsdomain touch genericstable makemap -r hash genericstable.db < genericstable mv sendmail.mc sendmail.mc.original wget http://nerdvittles.dreamhosters.com/pbxinaflash/source/sendmail/sendmail.mc.gmail cp sendmail.mc.gmail sendmail.mc mkdir -p auth chmod 700 auth cd auth echo AuthInfo:smtp.gmail.com \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" > client-info echo AuthInfo:smtp.gmail.com:587 \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" >> client-info echo AuthInfo:smtp.gmail.com:465 \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" >> client-info nano -w client-info
When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.
Now issue the following commands. In the last step, press ENTER to accept all of the default prompts:
chmod 600 client-info makemap -r hash client-info.db < client-info cd .. make sed -i 's|sendmail-cf|sendmail\/cf' /etc/mail/sendmail.mc sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/Makefile sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/databases sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc.gmail sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf.errors sendmailconfig
Next, edit /etc/hosts and /etc/hostname. Change the raspberypi3 entries to: raspberrypi3.incrediblepbx.com.
Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!
/etc/init.d/sendmail stop /etc/init.d/sendmail start apt-get install mailutils -y echo "test" | mail -s testmessage yourname@yourdomain.com
The last step is to add these commands to /etc/rc.local to send you an email with your IP address and SSID whenever the RasPi is rebooted. Insert the following commands just above the exit 0 line at the end of the file. Use an email address to which you have access on the road!
ESSID=`iwconfig | grep ESSID | tail -1 | cut -f 9 -d " "` echo "IP address: $(hostname -I) on $ESSID" | mail -s "RaspberryPi3 IP Address" yourname@yourdomain.com
Enabling Bluetooth on the Raspberry Pi
Implementing Bluetooth Proximity Detection with Smartphone and your #RaspberryPi3 https://t.co/TYFrcqcLux #asterisk pic.twitter.com/QqGNjqG3Ag
— Ward Mundy (@NerdUno) March 4, 2016
Incredible Fax Returns for the Raspberry Pi
PIONEERS: Free Faxing Returns for #RaspberryPi3 and Raspbian Jessie https://t.co/As1dMYPNLF #asterisk #IncrediblePBX pic.twitter.com/qDrJcy9cgn
— Ward Mundy (@NerdUno) March 7, 2016
Mastering the Incredible PBX Feature Set
Now would be a good time to explore the Incredible PBX applications. Continue reading there. 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. Enjoy!
Originally published: Monday, March 7, 2016 Updated: Saturday, March 26, 2016
Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a difficult place to address 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 Forum. 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, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won't have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
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