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The Most Versatile VoIP Provider: FREE PORTING

Dare to Compare: The Best (free) VoIP Offerings for 2018



Last week we showed you how to get 10 months of free hosting for your Incredible PBX® in the Cloud. And today we present our semi-annual survey of the latest and greatest VoIP offerings for 2018. The beauty of the cloud platform is you can try all of them for less than a penny an hour and decide for yourself which free offering best meets your needs. This year we’ve ushered in new Asterisk® 13 LTS releases of Incredible PBX® on the CentOS, Ubuntu, and Raspberry Pi platforms as well as new versions for Issabel 4 and VitalPBX. To sweeten the pot even further, we nailed down a new Cloud-based offering for $10 a year that makes a perfect VOIP sandbox for our CentOS platform. For 2018, we also secured new (free) DID offerings in the U.S. and announced a Nerd Vittles exclusive providing access to 300+ VoIP providers worldwide, all at wholesale prices. And, last but not least, we introduced Digium’s newest IP phones for Asterisk including a $59 model that makes a perfect VoIP companion.



Choosing the Best VoIP Platform for Your Needs

Choosing a VoIP platform is partially a subjective decision, but there also are some glaring red flags to consider. We suggest you begin by deciding whether your preferences include any must-have’s. Do your requirements mandate an open source solution? Do you need text-to-speech and voice recognition? Does the operating system have to be Linux-based and, if so, must it be CentOS, Debian, or Ubuntu? If you’ll be using SIP phones, must the platform include phone provisioning software for your phones, or is the ability to purchase it as an add-on sufficient? Is paid support important in making your platform decision and how much are you prepared to pay? Are automatic or pain-free software updates critical in making your selection? Is migration from an existing platform a factor? Does a preconfigured, secure firewall matter, or are you prepared to do it yourself or take your chances? Before choosing to ignore security, read this RIPS analysis of FreePBX®. Here’s a snippet from the article. Read it carefully. It’s your phone bill.

Since FreePBX is written completely in PHP, we decided to throw it into our code analysis tool RIPS. The results were more than surprising and should tell you why a rock-solid firewall is absolutely essential.

The total amount of detected vulnerabilities is very high. Luckily, the majority of the detected vulnerabilities are inside the administration control panel, such that attackers either need to steal a valid account or they have to trick an administrator into visiting a malicious website that triggers one of the critical vulnerabilities. For example, a remote command execution vulnerability could be triggered by a less critical cross-site scripting vulnerability. By chaining both vulnerabilities, the severity is increased drastically and can lead to full server compromise.

In choosing which platforms to include today, we eliminated platforms which we considered too complicated for the average new user to configure. We also eliminated any platform that did not offer at least a free tier of service with a reasonably complete feature set as part of their offering. So here’s our Pick of the Litter.

We must confess that we are partial to the Incredible PBX offerings because they provide a turnkey GPL platform with minimal configuration required on your part. Regardless of platform, all come standard with a preconfigured firewall and about three dozen applications for Asterisk that will help you learn everything there is to know about VoIP telephony.

VoIP Platform Feature Summary

Aggregation: Incredible PBX 13-13 for CentOS/SL
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: CentOS/SL 6.9 or 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers as well as ISO available

Aggregation: Incredible PBX 13-13 for Raspbian
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Raspbian 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure

Aggregation: Incredible PBX 13-13 for Ubuntu
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 13 GPL modules
O/S: Ubuntu 18.04
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic Update Utility included
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall Whitelist
Security Rating (as delivered): Secure
Comments: Lean & Mean or Whole Enchilada installers

Aggregation: VitalPBX
License: Closed Source
VoIP Platform: Asterisk 13
GUI: Free and Commercial modules
O/S: CentOS 7
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Incredible PBX add-on now available including TM3 firewall.

Aggregation: Incredible PBX for Issabel 4
License: Open Source GPL
VoIP Platform: Asterisk 13
GUI: FreePBX 11 GPL modules
O/S: CentOS 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Secure with Incredible PBX add-on
Comments: Incredible PBX add-on provides secure platform

Aggregation: FusionPBX for FreeSWITCH
License: Open Source MPL 1.1
VoIP Platform: FreeSWITCH 1.6
GUI: FusionPBX
O/S: Debian 8
Phone Provisioning: Free
Text-to-Speech/Voice Recognition: Optional/Optional
Software Updates: Automatic
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Secure with mods below
Comments: Incredible PBX firewall add-on now available .

Aggregation: Incredible PBX for Wazo
License: GPL3 Open Source
VoIP Platform: Asterisk 15 RealTime
GUI: Wazo GPL3 modules
O/S: Debian 9
Phone Provisioning: Extensive Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic or 2-minute Manual
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList with Incredible PBX add-on
Comments: High Availability & Call Center GPL3 Modules

Aggregation: FreePBX Distro a.k.a. AsteriskNOW
License: Closed Source
VoIP Platform: Asterisk 13/14/15
GUI: FreePBX GPL and Commercial modules
O/S: Closed-source CentOS fork
Phone Provisioning: Open Source (minimal) or Commercial
Text-to-Speech/Voice Recognition: Optional/No
Software Updates: Manual from Hidden Repo
Migration Tools: Yes
Security: Fail2Ban + User-Configurable Firewall
Security Rating (as delivered): Insecure
Comments: Extensive commercial NagWare preinstalled

 

Deploying a Local Server vs. Cloud Platform

We’ve always been big fans of local servers because you have almost total control of your own destiny. This was especially true when the Raspberry Pi came along to take the financial pain out of the server equation. But the price of Cloud-based servers has continued to plummet. For 2018, you can run any of our favorites on the least expensive platform at Vultr or Digital Ocean for $2.50 a month. And, if you hurry, your first 10 months are free at Vultr. Spending another 50 cents buys you automatic backups.1 And, for the Incredible PBX 13-13 build with CentOS 6.9 (64-bit), we’ve found a deal at HiFormance that offers a high-performance OpenVZ platform at an annual cost of just $10. The technical specs are impressive (even better if you sign up for 3 years), and we don’t think you’ll find a comparable deal with anything near comparable performance and specs anywhere, period. You get your choice of hosting sites including New York, Chicago, Los Angeles, Buffalo, Atlanta, and Dallas. Complete tutorial available here.

NOTE: OpenVZ/SolusVM platforms not suitable for CentOS 7, Debian 9, or Ubuntu 18 implementations, and some providers do not yet support Ubuntu 18.04 platform although Vultr and Digital Ocean both do.


Available Free Trunks for VoIP Servers

For many years, we’ve offered free Google Voice connectivity with our VoIP platforms. And that remains true at least for a few more weeks. On all of the Incredible PBX platforms, Google Voice trunks can be set up to make free calls in the U.S. and Canada provided you have a U.S. residence and a U.S. cellphone number to verify that you are who you say you are. There’s even a ray of hope that the Simonics gateway may allow you to continue using Google Voice after Google Voice’s mid-June drop-dead date for XMPP. Details here. But what about the rest of the world. For 2018, we solved the problem by offering free DID trunks for inbound calls and a collection of 300 wholesale VoIP carriers worldwide to make outbound calls at the same wholesale rates offered to the very largest resellers. Simply pay a 13% surcharge in lieu of the $650 annual fee, and TelecomsXchange (TCXC) will provide you access to their entire suite of wholesale carriers together with state-of-the-art tools to manage all of the services.2 The Nerd Vittles setup tutorial is available here. Enjoy!

Published: Monday, March 5, 2018  Updated: Sunday, May 27, 2018



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

  1. On the Vultr and Digital Ocean $2.50 platforms, be sure to (1) create a 1GB swapfile once you’ve chosen your operating system. (2) Then, for Vultr, issue the following command before beginning the Incredible PBX install: apt-get install cloud-init.
    (3) Now complete the steps outlined in your preferred Nerd Vittles tutorial, and you’ll be all set in about 15 minutes. []
  2. Our special thanks to TelecomsXchange. They have generously offered to contribute a portion of the wholesale surcharge to support the Incredible PBX open source project. []

Revolutionary VoIP: The Best (free) PBX Ever from 3CX

There are evolutions, and then there are revolutions. Today is another revolutionary day for free VoIP. The new 3CX v15.5 Update 3 is revolutionary on so many levels: price, feature set, flexibility, stability, and security for openers. For Nerd Vittles readers that want a free PBX for your home or business, here’s the latest and greatest. You get the 3CX Standard License features listed here with up to 16 simultaneous calls for one year. That setup easily supports about 50 extensions. At the expiration of the year, you can purchase the standard annual license OR your free license will automatically convert to a 4-simultaneous-call perpetual license with unlimited trunks for the duration of the installation, including DNS, email, SSL certs, webmeeting, etc. Nothing else to buy ever!1 This perpetual license includes unlimited SIP trunks and gateways, 25-participant conferencing, G.722 and G.729 support with HD Voice, custom FQDNs, BLF support, Call Parking, Call Queueing, Call Pickup, Call Recordings and Management, Call Reporting, Intercom/Paging, Integrated Fax Server and Office 365 Address Book/Microsoft Outlook integration plus all of the 3CX client software. Better hurry. This offer won’t last forever! Here’s the signup link. 2

Unlimited Trunks, 50 Extensions, 16 Simultaneous Calls… Free!

The 3CX development team not only heard but also heeded our suggestion to expand the number of trunks in the free edition by removing the limitation entirely. With small businesses and home users, the number of times you ever will need to make more than 16 simultaneous calls is probably NEVER. Based upon industry standards, this 16-call, 50-extension PBX with unlimited trunks can easily support several dozen people so it’s perfect for home use and small to medium-sized businesses. And, when your business grows, upgrading to a larger PBX is inexpensive and a one-minute key swap.

Cost savings, of course, are only part of the VoIP story. There’s a reason 3CX’s business is growing geometrically while others struggle. 3CX provides an unmatched feature set that’s easy to use and deploy. Version 15.5 Update 3 brings the Linux platform to full parity with 3CX’s previous Windows editions plus all-new 3CX clients for every desktop and mobile device. There’s also an awesome new web client providing users easy access to all key 3CX features without installing any software. Desktop call control including Click2Call now is based on uaCSTA technology. Snom, Yealink, and Granstream phones as well as 3CX clients can be controlled from any desktop client even if your phone system is running in the cloud. And we’ve got a whopper deal for you there as well today.

With 3CX’s powerful client software, your office and your PBX can literally be anywhere. Your desktop is always as close as your smartphone or the nearest WiFi hotspot. That’s what unified communications is all about. And, should you ever need support, 3CX has offices in the U.S., U.K., Germany, Hong Kong, South Africa, Russia and Australia. Review the 3CX feature comparison chart and you can judge the feature set for yourself. Whether you’re a homebody or world traveler, we think you’ll agree that 3CX’s new free edition for Nerd Vittles readers offers everything that a home or SOHO user will ever need in a PBX.

Getting Started with 3CX on Dedicated Hardware or a Virtual Machine. If your platform supports ISO installs, here are the simple steps to get 3CX up and running. Just follow this 3CX tutorial to download the ISO and begin your adventure. Boot your server from the ISO image and walk through the Debian 9 setup process. We recommend 2GB of RAM and a 20GB drive for 3CX. When the install is finished, make note of the IP address to access with a web browser to complete the setup. Enter your 3CX license key when prompted. Set up one or more SIP trunks with inbound and outbound call routes. Once you have the ISO and your license key in hand, the installation procedure takes less than 10 minutes.

Getting Started with 3CX in the Cloud. Begin by setting up a 64-bit Debian 9 platform. Obtain a free Nerd Vittles license key for 3CX. Once your Debian install is finished, log in as root using SSH or Putty and issue these commands. NOTE: What appears as the third line below needs to be added to line #2!

wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -
echo "deb http://downloads.3cx.com/downloads/debian stretch main" | tee /etc/apt/sources.list.d/3cxpbx.list
apt-get update
apt-get install libcurl3=7.38.0-4+deb8u5
apt-get install net-tools
apt-get install 3cxpbx

When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Set up one or more SIP trunks with inbound and outbound call routes. Done.

Beginning with this release, you have your choice of using a Google Cloud-hosted 3CX server at no cost for a year or many other cloud providers of your choice. The problem with the Google Cloud offering is what to do after the first year. Our personal preference is to set up your own cloud server where things stay the same as you move forward from year to year. At this time, 3CX does not support OpenVZ containers. However, Vultr offers a $2.50/month 512MB RAM plan that works just fine. 50 cents more buys you automatic backups that we highly recommend. And OVH offers quadruple the RAM for $4.49/month on a 12-month plan.

Configuring Gmail as SMTP RelayHost for 3CX. 3CX has a detailed tutorial explaining how to set up your Gmail account as the SMTP relay host for 3CX. Be advised that there is one additional step before Google will authorize access from an IP address it doesn’t already have for your GMail account. In addition to Enabling Less Secure Apps (as covered in the 3CX tutorial), you also will need to activate the Google Reset Procedure while logged into your Gmail account. Otherwise, Google will block access. Once you have configured Gmail as your relay host and performed the two enabling steps above, immediately test email delivery within the 3CX GUI while Google security is relaxed: Settings → Email → TEST.

Free Calling in the U.S. and Canada with 3CX. We know our more frugal U.S. residents are wondering if there’s a way to make free calls even with 3CX. You didn’t really think there would be a release of PBX in a Flash without Google Voice support, did you? It’s easy using the Simonics SIP to Google Voice gateway service. Setup time is about a minute, and the one-time cost is $4.99 using this Nerd Vittles link. Setup instructions for the 3CX side are straight-forward as well, and we’ve documented the procedure on the PIAF Forum.

Free Calling Worldwide with SIP URIs. There’s another free calling option as well. 3CX supports worldwide SIP URI calling at no cost. As part of the 3CX install procedure, 3CX registers an FQDN for you with one of the 3CX domains if you indicate that your server has a dynamic IP address. Unless you really know what you’re doing with DNS, it’s a good idea to tell 3CX you have a dynamic IP address whether you do or not. Here’s why. Once you have an assigned FQDN in the 3CX universe, one very slick feature is the ease with which you can publish a SIP URI address for any or all of your 3CX extensions thereby allowing 3CX users to receive calls from any SIP client worldwide at no cost. Setup takes less than a minute. It’s as easy as 1-2-3. Here’s how:

1. Login to the 3CX GUI and go to Settings → Network → FQDN. Tick "Allow calls from/to external SIP URIs" and make note of your FQDN, e.g. mypiaf5server.3cx.us. Click OK.

2. For an extension to enable (e.g. 001), go to Extensions → Edit 001 → Options → SIP ID and create any desired SIP URI alias for this extension, e.g. billybob. Click OK.

3. If your PBX is sitting behind a router/firewall, be sure the following UDP ports are forwarded to the local IP address of your PBX: 5001, 5060, 5090, and 9000-9255.

4. Anyone with a SIP client anywhere worldwide can now call extension 001 using SIP URI: billybob@mypiaf5server.3cx.us.

Originally published: Wednesday, June 7, 2017  Updated: Thursday, February 8, 2018



Need help with 3CX or VoIP? Visit the PBX in a Flash Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

  1. This offering applies to 3CX V15.5 Update 3 released on February 8, 2018. []
  2. Don’t confuse 3CX’s free PBX with Sangoma’s FreePBX® GUI. The former is a truly free PBX provided by a well-respected developer of commercial PBXs and used by many of the world’s largest companies including Boeing, McDonalds, Hugo Boss, Ramada Plaza Antwerp, Harley Davidson, Wilson Sporting Goods, and Pepsi. The latter is a code generator for Asterisk® that commingles free components with commercial NagWare, each of which requires payment of separate licensing and maintenance fees before and during subsequent use. []

The World Traveler and 3CX: A Match Made in Heaven

Last week we introduced the new (free) version of PIAF5 powered by 3CX v15.5 supporting four simultaneous calls, unlimited trunks, 10 extensions, and 10-user conference calls. And today we’re torture-testing our new 3CX server in the Bahamas aboard one of Carnival’s 3,000-passenger floating cities. Somebody’s gotta do it, right? What makes this such a challenging test for any PBX are several things. First, we’re using a free Google Voice trunk on a free 3CX PBX that we configured in under 10 minutes at CloudAtCost for a one-time cloud server charge of $17.50. Second, we’re sharing a satellite Internet connection with 3,000 other people in the middle of the Caribbean. The weekly charge is about $100 so every Internet junkie subscribes. Third, we’re using a 3CX Client on an iPhone in Airplane Mode. And, finally, we’re sitting behind the most Draconian firewall you can imagine because Carnival assumes everyone is a bad guy trying to bring their Internet service to its knees.

For those coming from the Asterisk® world, I don’t have to remind you how challenging this NAT-based setup would be even assuming you had a flawless Internet connection. Believe me. We don’t. And the secret sauce that makes all of this seem like child’s play is the latest collection of 3CX Clients for PCs, Macs, Android devices, and iPhones/iPads. Simply download the client for your platform, log into your 3CX portal and send the welcome email from a configured extension to your phone, open the email on your phone and double-click on the attachment, and boom. Your 3CX Client is automatically configured in seconds and ready to make your first call. A monkey could do it. It’s that easy!

So our torture-test for today looks more like a final exam in VoIP telephony. We’ll be using Carnival’s WiFi connection from our iPhone with its iOS 3CX Client. We’ll dial into the Incredible PBX™ at our office in Charleston. The office number is configured with a Stealth AutoAttendant which we’ll use to make an outbound call to our Demo IVR in Marbella, Spain using DISA and a FreeVoipDeal trunk. For the techies, it’s the NAT Trifecta with DTMF hurdles that are virtually impossible to traverse using Asterisk and any SIP client.

Guess what? It not only works, but it sounds like you’re sitting in the adjoining office. No echo, no DTMF problems, no missing audio, and no detectable problems in voice quality with either the Charleston IVR or the Marbella IVR. If cost matters and traveling is a key component in your telephony requirements, you owe it to yourself to set up a free 3CX PBX and take it for a spin. Whether you use it to supplement an existing Asterisk setup or as a standalone PBX, we think you’ll be thrilled with the results.

Continue reading about the new, free PIAF5 server powered by 3CX v.15.5

Originally published: Monday, June 12, 2017



Need help with 3CX or VoIP? 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…

Best of Both Worlds: Marrying Asterisk to 3CX’s Free PBX with a $35 Raspberry Pi


One of the real beauties of Asterisk® has always been its flexibility in talking to other PBXs, both commercial and open source. There are numerous reasons why you might want to try this. First, it makes it easy to migrate to a commercial platform where you can get support for mission critical telephony requirements. Second, you may want a hybrid setup where servers with on-site support personnel can run Asterisk while remote satellite offices can take advantage of a commercial PBX and the support options it offers. Third, you may want to take advantage of specific features that are only available by relying upon multiple PBX solutions. In the case of 3CX, their integrated softphone clients with one-click setup simplicity, conferencing and WebRTC apps, and Call Center offerings are the best in the business while providing unmatched VoIP security. Asterisk on the other hand is light-years ahead of almost everybody in the text-to-speech and voice recognition fields while offering the most powerful VoIP toolkit to build any custom VoIP application imaginable.

Today we thought it would be fun to walk you through the easy way to tie an Incredible PBX server with all its features to a powerful (free) 3CX platform with its virtually flawless softphone clients.1 When we’re finished, you’ll have a free 3CX server in the Cloud at a one-time total cost of $17.50. And you’ll be able to place and receive free U.S./Canada calls from any iPhone, Android phone, or PC using the 3CX client from anywhere in the world with nothing more than a WiFi connection. The Google Voice trunk supporting the calls will reside on Incredible PBX for the Raspberry Pi. When you’re sold on the power of the 3CX platform, you can upgrade to the 3CX 4-simultaneous call commercial offering with unlimited users and trunks at an annual cost of just $149. Maintenance and upgrades are included. Large organizations have relied upon back office servers for custom applications forever. And now you can take advantage of the same flexibility using a tiny $35 Raspberry Pi and our free (as in really free) Incredible PBX software. No Gotchas!

Initial Raspberry Pi Platform Setup

Before we can interconnect 3CX’s Free PBX with a Raspberry Pi, you obviously have to set up both PBX platforms. For the Raspberry Pi, our recent Nerd Vittles tutorial will walk you through the setup process. In lieu of a Raspberry Pi, you can use any legacy FreePBX®-based Asterisk platform including Incredible PBX 13, PIAF3, Elastix®, AsteriskNOW®, or FreePBX Distro®. The setup procedure is exactly the same.

Building a 3CX Server in the Cloud

Building a 3CX server in the Cloud is equally easy. Let’s go through the process once again. If you’re just experimenting, a lifetime Cloud-based server at CloudAtCost for a one-time charge of $17.50 cannot be beat. We would hasten to add that we don’t recommend this platform for production use, but it’s a terrific proof-of-concept option. When you’re actually ready to deploy 3CX for production use, the least costly Cloud solution is the $3.49 per month OVH RAID offering with 2GB of RAM and 10GB storage. The $5 per month offerings from Digital Ocean and Vultr are other alternatives worth a look. Both of these platforms come with free credits ($10 and $20, respectively) to let you try things out.

To get started, sign up for a $17.50 server at Cloud at Cost. They will send you credentials to log into the Cloud at Cost Management Portal. Change your password IMMEDIATELY after logging in. Just go to SETTINGS and follow your nose.

To build your free 3CX PBX, create a virtual machine by clicking on the CLOUDPRO button in the CloudAtCost control panel. Then click Add New Server. Choose 1 CPU, 512MB RAM, and 10GB storage for your server. Choose Debian 8 64bit as the OS Type and click Complete.

While CloudAtCost is building your server platform, obtain a free license key for 3CX.

Once the Debian 8 server appears in your Control Panel, it will look something like what’s shown above, not CentOS obviously. The red arrow points to the i button you’ll need to click to decipher the password for your new virtual machine. You’ll need both the IP address and the password for your new virtual machine in order to log into the server which is now up and running with a barebones Debian 8 operating system. Note the yellow caution flag. That’s telling you that Cloud at Cost will automatically shut down your server in a week to save (them) computing resources. You can change the setting to keep your server running 24/7. Click Modify, Change Run Mode, and select Normal – Leave Powered On. Click Continue and OK to save your new settings.

Finally, you’ll want to change the Host Name for your server to something more descriptive than c7…cloudpro.92… Click the Modify button again and click Rename Server to make the change. Your management portal then will show the new server name as shown above.

Next, log in to your new Debian server as root using SSH or Putty and issue the commands below. Step #1 is to change your root password. What appears as the fourth line below is actually part of the third line and needs to be run as a single command. The last line to install SendMail will actually be run after you elect to use the Web Interface Wizard to configure 3CX. Just run it from the SSH command line before you switch to a browser to complete the 3CX setup.

passwd
wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -
echo "deb http://downloads.3cx.com/downloads/3cxpbx/ /" | tee /etc/apt/sources.list.d/3cxpbx.list
apt-get update
rm -f /zang-debian.sh
apt-get -y install 3cxpbx
apt-get -y install sendmail sendmail-bin

When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Make up a very secure Username and Password to access your 3CX portal. Specify that your IP address is Dynamic when prompted (even though it isn’t). This tells 3CX to generate an FQDN for your server. Accept the default ports for HTTP (5000) and HTTPS (5001) access to your server. We recommend choosing 4-digit extensions numbers which will make it easy to distinguish 3CX extension numbers from 3-digit extension numbers of the RasPi platform. While logged into the 3CX management portal, adjust Settings → Email to Mail Server → 127.0.0.1 and Reply to → noreply@YourActual3CX-FQDN. Leave the other settings blank and click TEST then OK. Now download your favorite 3CX smartphone client, send yourself the Welcome Email for your default extension, and your 3CX initial setup is complete.

Server Interconnection Overview

Now we’re ready to interconnect the two servers. What we’ll be doing is creating Trunks on both the Raspberry Pi and the 3CX server and tying them together. We’ll use this trunk to handle the call traffic between the two PBXs. Then we’ll add incoming and outgoing call routes on both servers to specify how the individual calls should be routed. Because the free version of 3CX limits the administrator to a single trunk, we’ll offload all of the provider trunks to the Raspberry Pi and reserve the one available 3CX trunk as the interconnect path to the Raspberry Pi. For today’s setup, we’ll use 3CX’s free softphone clients as the actual phone devices for end-users. Of course, you could also use your favorite SIP phones, and 3CX provides automatic configuration for dozens of devices. But we want to introduce the 3CX smartphone clients because they provide an incredibly easy way to get users connected without having to worry about punching holes in firewalls.

To place outbound calls on the 3CX side, 3CX provides enormous flexibility in call routing. Because we chose 4-digit local extensions when we set up the 3CX server, it will make it easy to route other calls through the outbound trunk to the Raspberry Pi using nothing more than the length of the dial string. For example, 3-digit calls line up perfectly with extension numbers on the Incredible PBX for RasPi platform. So 3CX users can easily reach extensions connected directly to the Raspberry Pi. And 10-digit 3CX calls will be forwarded to the Raspberry Pi as traditional outbound calls. They will be processed just as if you had dialed a 10-digit call from a Raspberry Pi extension. For example, if you have a registered Google Voice trunk to handle 10-digit calls on the Raspberry Pi, then the same call path would be used for calls originating from 3CX extensions. And, yes, calls to the U.S. and Canada would still be free and would display the CallerID associated with the Raspberry Pi’s Google Voice trunk. You could get more creative and add an additional dialing prefix on the 3CX side to route specific types of calls to a designated outbound trunk on the Raspberry Pi side based upon the dialing prefix, but we’ll leave that as a homework project for you.

For incoming calls on the 3CX side, in addition to 4-digit local extension-to-extension calling, we can define the destination for incoming calls that originate from either a Raspberry Pi extension or from outside calls coming in from one of the Raspberry Pi’s provider trunks. These are managed by assigning one or more DIDs in the 3CX trunk configuration and then creating 3CX Inbound DID Rules that tell 3CX where to route calls to each defined DID. For 3CX softphone clients registered to extensions, it means your cellphone will ring whenever a call is routed to that particular extension. On the Raspberry Pi side, we create Incoming Call Routes for each DID to be routed to 3CX and specify our defined 3CX trunk as the destination for incoming calls from those DIDs. Not all DIDs on the Raspberry Pi have to be routed to the 3CX server obviously. That is merely one of many call destination options available to the administrator on the Raspberry Pi server.

Here’s a typical call path for an outside call that is placed to a Google Voice number registered with your Raspberry Pi. The Asterisk server running on the Raspberry Pi would answer the call placed to the Google Voice Trunk. Asterisk then would check for an Incoming Route on the Raspberry Pi with a DID matching the number of your Google Voice trunk. Finding a match, Asterisk would check for the desired destination of the call and would note that it is listed as the registered 3CX trunk. Asterisk would pass the call through this trunk to the 3CX server including its associated DID and CallerID info. The 3CX server would answer the incoming call and would check for an Incoming Route matching the DID passed from Asterisk. Finding a match, it would pass the call to the Extension specified in the Incoming Route. When 3CX rings the extension, it would also detect that a softphone was registered to that extension and would also ring the 3CX client on the user’s smartphone. The user answers the call on the 3CX client of their smartphone and begins a conversation. The free version of the 3CX server supports 8 simultaneous calls so you are unlikely to ever run out of call paths for calls in the home and small office environment.

Firewall Setup for Server Interconnection

Because the 3CX server is sitting in the Cloud, its firewall is configured automatically as part of the setup process. If your Raspberry Pi is sitting behind a NAT-based firewall, then you would need to map port UDP 5060 from the router on your public IP address to the private IP address of your Raspberry Pi. In addition, login to your Raspberry Pi as root using SSH and run /root/add-ip to whitelist the public IP address of your 3CX server in the cloud. Otherwise, the 3CX server cannot establish a connection to your Raspberry Pi.

Raspberry Pi Trunk Configuration

Using a browser, login to the web interface for FreePBX on your Raspberry Pi and choose Connectivity → Trunks → Add SIP (chan_sip) Trunk. Name the trunk remote. In the Outgoing Settings, make the entries shown below naming the trunk remote and using a secure secret that will be used to interconnect the two servers. The Register String looks like the following: main:secret@3CX-IP-Address where main is the 3CX server trunk name, secret is your secure secret, and 3CX-IP-Address is the 3CX public IP address.

3CX Trunk Configuration

Using a browser, login to your 3CX server: https://3CX-IP-Address:5001 or http://3CX-IP-Address:5000. From your Dashboard, choose SIP Trunks → Add SIP Trunk. Create a Generic SIP Trunk and then fill in the blanks as shown below. For Registrar/Server/Gateway Hostname or IP, use the public IP address or FQDN of your Raspberry Pi. For Type of Authentication choose Outbound. The authentication credentials should be remote and the secure secret you chose, and the Main Trunk No should match the DID of the Google Voice trunk you set up on your Raspberry Pi. Then pick a default Destination for incoming calls.

3CX Outbound Rules Configuration

Next, we need to tell 3CX which outgoing calls to send out through the Raspberry Pi trunk we just set up. In our example today, we’re going to send all 10-digit calls and 3-digit calls. The 10-digit calls will be routed out the Google Voice trunk on the Raspberry Pi side. And the 3-digit calls will be sent directly to Raspberry Pi extensions. So we’ll need two Outbound Rules.

For the first rule, choose Outbound Rules → Add. For the Rule Name, specify StandardOut. Apply the rule to Calls to Numbers with a length: 10. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.

For the second rule, choose Outbound Rules → Add. For Rule Name, specify StandardInt. Apply the rule to Calls to Numbers with a length: 3. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.

If you already have configured a 3CX smartphone client for one of your 3CX extensions, you now should be able to dial any 3-digit or 10-digit number and have the call processed through your new 3CX→RasPi trunk without any further setup assuming you’ve created a Google Voice trunk on the Raspberry Pi side. That wasn’t too hard, was it?

Routing Incoming Google Voice Calls to 3CX

Depending upon your own requirements, you may want to route incoming Google Voice calls or other trunks directly to an extension and/or softphone on your 3CX server. You obviously could set up multiple trunks of any type on the Raspberry Pi side and have the calls to each trunk routed to a different extension or softphone on the 3CX side. To enable this on the 3CX side, edit your Generic SIP Trunk and click the DIDs tab. Then Add each of the 10-digit DIDs of the Raspberry Pi trunks you wish to redirect. Next, create an Inbound Rule for every DID and tell 3CX where to route the calls.

On the Raspberry Pi side, add each of your Google Voice Trunks. Then create an Inbound Route for each DID and specify the Destination as Trunks → Remote (sip). The 3CX server will take care of routing the various incoming calls to each of the Google Voice trunks to its predefined extension and/or softphone. Enjoy!

Originally published: Monday, March 6, 2017





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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

  1. A simpler Bridge setup is available in the paid versions of 3CX. []

Free At Last: Introducing PBX in a Flash 5



Today is a big day. We are thrilled to introduce PBX in a Flash 5 powered by 3CX®. As many of you know, 3CX has been a platinum sponsor of Nerd Vittles for quite some time so this may not be a complete surprise. The good news is a new Debian-based PIAF5 ISO is now available to ease the installation process for those getting their feet wet with Linux for the first time. Debian 8 is a terrific Linux distribution used in the very best server products.

The most important change is the transition from Asterisk®/FreePBX® to 3CX. Say what, 3CX? Isn’t that a commercial product? Yes, but PIAF5 remains free for up to 8 simultaneous calls with a SIP trunk as well as 5-user web conferencing. That’s sufficient to support about 25 employees and represents a very large segment of the existing PIAF installed base. While the code is not open source, it is standards-based. Keep in mind that neither Sangoma’s FreePBX Distro® nor Digium’s AsteriskNOW® product is open source software either. When Digium decided to adopt the Sangoma business model, we decided to take a fresh look at the Unified Communications landscape. Navigating Sangoma’s licensing labyrinth coupled with the commingling of GPL modules and nagware for dozens of commercial VoIP components plus a closed source ISO was no longer an acceptable business model for us.

Some of our users prefer open source code, and we will continue to enhance Incredible PBX for XiVO in the grandest GPL tradition. But others wanted a product that offered 24×7 commercial support, and we’ve heard you loud and clear. After carefully reviewing available UC offerings, 3CX was the hands down winner in the commercial sector. Frankly, our only reservation was its Windows platform requirement. PIAF5’s new Debian ISO solves that.

In reality, what matters to users are reliability, support, upgradeability, and ease of use. 3CX has all of them in spades not to mention a feature set that is second to none. And now it’s available on the Debian platform with PIAF5.

We know some are wondering how 3CX became the new PIAF5 platform. So let’s start there.

First, the 3CX installed base includes almost 100,000 companies. That’s not downloads. And it’s not hobbyists. It’s entire companies that are actively using and relying upon 3CX for their day-to-day operations. Simply stated, 3CX is a proven, stable, and dependable product that you’d be willing to stake your business on. Many have including some of the world’s finest corporations. Stay tuned for a special PIAF5 hosting offer from our friends at Vitelity!

Second, 3CX is incredibly flexible, easy to configure, and simple to manage. Whether you’re new to PBXs or a diehard telecom guy, you’re in for a pleasant surprise when you see how intuitive 3CX is to set up and manage. Nothing comes close in the open source world.

Third, the 3CX feature set is impressive. You won’t be nickel and dimed for every component you wish to add. While there are standard and enterprise editions of 3CX as well, we think you’ll find the free version has the vast majority of components you would expect to find in any PBX, particularly for use in a home or small business. But don’t take our word for it. Review the 3CX feature comparison chart, and you can judge for yourself.

Last but not least, support is dirt cheap for end-users and free for resellers. We hope many of our long-time gurus will consider signing up as 3CX resellers and make yourself some money after all of these years wrestling with FreePBX. You won’t be disappointed!

PIAF5 deploys on premise with Linux-compatible, local hardware, or you can set it up as a virtual machine, or you can install it in the Cloud using most Linux VPS providers including Google, OVH, Digital Ocean, and Vultr. Use our referral links and take PIAF5 for a free or almost free spin for a few months while supporting Nerd Vittles. You have nothing to lose!

So there you have it. We think it was worth the wait. We encourage everyone to try out PIAF5 for yourself. And, just to repeat, Incredible PBX for XiVO isn’t going anywhere. It will remain our featured open source, GPL alternative as we move forward. And now you have a Real Choice in free alternatives with the best of both worlds, commercial and open source.

Getting Started with PIAF5 on Dedicated Hardware or a Virtual Machine. If your platform supports ISO installs, here are the simple steps to get PIAF5 up and running. First, download the PIAF5 ISO and burn it to a CD or thumb drive. Second, obtain a free license key for 3CX. Next, boot your server from the ISO image and walk through the Debian setup process. We recommend 2GB of RAM and a 20GB drive for PIAF5, but it will run on even a minimal CloudAtCost server. When the install is finished, make note of the IP address to access with a web browser to complete the setup. Enter your 3CX license key when prompted. Set up a SIP trunk with inbound and outbound call routes. Once you have the ISO and your license key in hand, the installation procedure takes less than 10 minutes.

Getting Started with PIAF5 in the Cloud. Begin by setting up a 64-bit Debian 8 platform. Obtain a free license key for 3CX. Once your Debian install is finished, log in as root using SSH or Putty and issue these commands. NOTE: What appears as the third line below needs to be added to line #2!

wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add -
echo "deb http://downloads.3cx.com/downloads/3cxpbx/ /" | tee /etc/apt/sources.list.d/3cxpbx.list
apt-get update
apt-get install 3cxpbx

When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Set up a SIP trunk with inbound and outbound call routes. Done.

Configuring Gmail as SMTP RelayHost for 3CX. 3CX has a detailed tutorial explaining how to set up your Gmail account as the SMTP relay host for 3CX. Be advised that there is one additional step before Google will authorize access from an IP address it doesn’t already have for your GMail account. In addition to Enabling Less Secure Apps (as covered in the 3CX tutorial), you also will need to activate the Google Reset Procedure while logged into your Gmail account. Otherwise, Google will block access. Once you have configured Gmail as your relay host and performed the two enabling steps above, immediately test email delivery within the 3CX GUI while Google security is relaxed: Settings → Email → TEST.

Free Calling in the U.S. and Canada with PIAF5. We know our more frugal U.S. residents are wondering if there’s a way to make free calls even with 3CX. You didn’t really think there would be a release of PBX in a Flash without Google Voice support, did you? It’s easy using the Simonics SIP to Google Voice gateway service. Setup time is about a minute, and the one-time cost is $4.99 using this Nerd Vittles link. Setup instructions for the 3CX side are straight-forward as well, and we’ve documented the procedure on the PIAF Forum.

Free Calling Worldwide with SIP URIs. There’s another free calling option as well. PIAF5 and 3CX support worldwide SIP URI calling at no cost. As part of the PIAF5 install procedure, 3CX registers an FQDN for you with one of the 3CX domains if you indicate that your server has a dynamic IP address. Unless you really know what you’re doing with DNS, it’s a good idea to tell 3CX you have a dynamic IP address whether you do or not. Here’s why. Once you have an assigned FQDN in the 3CX universe, one very slick feature is the ease with which you can publish a SIP URI address for any or all of your 3CX extensions thereby allowing PIAF5 users to receive calls from any SIP client worldwide at no cost. Setup takes less than a minute. It’s as easy as 1-2-3. Here’s how:

1. Login to the 3CX GUI and go to Settings → Network → FQDN. Tick "Allow calls from/to external SIP URIs" and make note of your FQDN, e.g. mypiaf5server.3cx.us. Click OK.

2. For an extension to enable (e.g. 001), go to Extensions → Edit 001 → Options → SIP ID and create any desired SIP URI alias for this extension, e.g. billybob. Click OK.

3. Anyone with a SIP client anywhere worldwide can now call extension 001 using SIP URI: billybob@mypiaf5server.3cx.us.

SMS Messaging with PIAF5 and Google Voice. Just to demonstrate why you’re going to love the new PIAF5 platform, here’s a sneak peek at one of many applications which are on the way with Incredible PBX for PIAF5. Meet SMS Messaging. First, complete the two Google enabling steps documented in the Gmail SMTP RelayHost section above: Enable Less Secure Apps and Activate Google Reset Procedure. Then install the Google Voice CLI tools as root:

cd /root
apt-get -y install python-setuptools
wget http://incrediblepbx.com/install-gv-cli
chmod +x install-gv-cli
./install-gv-cli

To Send an SMS Message Blast to one or more destinations, (1) create a message in /root/smsmsg.txt, (2) specify the SMS numbers in /root/smslist.txt, (3) insert your Google credentials into /root/smsblast, and (4) run /root/smsblast to send the message. Enjoy!

Published: Wednesday, October 19, 2016




 

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.


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.





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, 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!


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





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…

VoIP Hardware Deal of the Year: Meet the $20 Pogoplug 4 with Incredible PBX

This week’s project is not for mere mortals. It’s for techies that also happen to be cheapskates frugal. You may recall the Pogoplug from yesteryear. Well, the Pogoplug 4 still is around and can be yours for under $20 with free 2-day shipping if you’re an Amazon Prime member. But the clock is ticking on these bad boys. Once they’re gone, they’re gone.1

UPDATE: There’s more good news. Now the cost with the Pogoplug Backup & Sharing model is just $10.95! For our purposes, the main difference is one less USB port, but it still has one which is all you need for wireless networking.

So we took the dare and decided to see whether the Pogoplug 4 could actually run Asterisk 11® and FreePBX® 2.11 and Incredible PBX™. And guess what? It may not be pretty, but it works. If you happen to have a Google Voice number and a kid away at school or a grandma in a distant city with an Internet connection or if you have a vacation home or rental property that needs phone service (but not often), then this $20 project may be for you. Configure the device, add a cheap SIP phone, and presto! You’ve got free calling in the U.S. and Canada with your very own phone number for as long as you have Internet service and Google chooses to keep paying your phone bill. 😉

Don’t take our word for it. Call our Pogoplug for a quick IVR demo compliments of Allison:

  1. Call by Name (just say "American Airlines" to try it out)
  2. Conference Call (enter 1234# to join the conference)
  3. Wolfram Alpha (try this: "What planes are overhead?")
  4. Lenny (the Telemarketers’ Worst Nightmare)
  5. Yahoo News Headlines
  6. Weather Forecasts (say a city and state or country)
  7. Today in History
  8. Ring the House Phones (sends you back to Lenny)

Our tip of the hat this week goes to Qui Hong without whom none of this would have been possible. His tutorial on transforming the Pogoplug 4 into a Debian server is a true masterpiece. And his blog is where we begin our adventure once you have the correct Pogoplug 4 in hand: POGO-V4-A3-01. Our link has the correct one, but double-check the Model Number just to be sure.

Converting the Pogoplug 4 into a Debian Platform

Once you have your Pogoplug, you’ll need to scurry over to Qui Hong’s blog and carefully work through his tutorial to convert your Pogoplug into a Debian server. As we’ve said many times before, if you can follow a cookie recipe and end up with edible cookies, then you can do this. Just be very careful of typos. One bad keystroke can turn your Pogoplug into a burnt cookie. Then it becomes a $40 project. 🙂

Installing Incredible PBX 11.12.0 on the Pogoplug 4

Once your Pogoplug has been Debianized, there are five simple steps to get Incredible PBX up and running on your Pogoplug 4:

  1. Purchase a storage device
  2. Download Incredible PBX image
  3. Untar the image on your desktop
  4. Burn the image to an SD card
  5. Insert the SD card in the Pogoplug and boot

Choosing a Storage Platform. The first step is to purchase a suitable SD card. We recommend at least a 16GB Class 10 card from Transcend, SanDisk, or Kingston. All of them are about $10 on Amazon and many include free 2-day shipping for Prime customers.

Downloading Incredible PBX for Pogoplug. From your favorite desktop computer, download the latest build of Incredible PBX from SourceForge. Depending upon your network connection and the SourceForge mirror, it can take awhile. It’s a whopping 1.5GB image!

Untarring Incredible PBX for Pogoplug. Depending upon your desktop platform, untarring incrediblepbx.4.pogoplug.D7.latest.tar.gz is as simple as double-clicking on it in the Downloads folder (on a Mac). On the Windows platform, here are 3 utilities that will do the job. On a Linux desktop, open a Terminal window and…

tar zxvf incrediblepbx.4.pogoplug.D7.latest.tar.gz

Burning the Incredible PBX image to SD card. Once you’ve untarred the file, you’ll find two scripts that make burning the image to an SD card simple if you’re on a Mac or Linux desktop. On a Windows machine, it’s a little more complicated. Most SD cards come preformatted with a DOS partition so your Windows machine should recognize the SD card when it’s inserted. If not, format the card using a utility such as SD Card Formatter. Next, you’ll need Win32 Disk Imager to burn pogoplug.img to your card. Once the image has been transferred, gracefully unmount the card from your desktop.

Booting Incredible PBX on the Pogoplug. Insert the SD card (electronics side down) into your Pogoplug 4. Then apply power to the device after connecting an Ethernet cable to a network with Internet connectivity that can also hand out DHCP addresses. Visit your router to decipher the IP address assigned to the Pogoplug and reserve the IP address so that it doesn’t suddenly change down the road. Log into Incredible PBX as root with pogoplug as your password. Your SSH credentials, Asterisk DUNDI secrets, logs, and network connection options will be initialized. When prompted, press Enter to reboot your server. With some SD cards, you may find yourself waiting an eternity for the promised reboot. After seeing the "rebooting" message, count to ten. If your server still hasn’t rebooted, remove and reapply power. This quirk goes away after the first reboot.

After the reboot, log in again as root with password: pogoplug. Your firewall setup will be initialized to lock down your whitelist to your server’s public and private IP addresses AND the IP address of the machine from which you’re logging in. All of your FreePBX passwords will be randomized as well. The whole process only takes a few seconds.

When the second pass configuration is complete, you will be greeted by a welcoming message. STOP and read it. It has loads of important information about your server’s configuration and your next steps. Press ENTER to review status:

The Next 10 Steps. Before you do anything else, complete the following steps. It only takes a minute to secure and properly configure your server:

  1. Change your root password: passwd
  2. Change your FreePBX admin password: /root/admin-pw-change
  3. Set your correct time zone: /root/timezone-setup
  4. Expand partition to match SD card size: /root/resize-partition
  5. Add any desired IP addresses to WhiteList: /root/add-ip
  6. Decipher the randomized password for extension 701. It’s in the data field:
    mysql -uroot -ppassw0rd -e "select * from asterisk.sip where id=701 and keyword='secret'"
    
  7. Decipher the randomized voicemail password for extension 701. It’s the first entry:
    cat /etc/asterisk/voicemail.conf | grep 701 | cut -f 3 -d " "
    
  8. Enable Windows Networking, if desired: /root/samba-enable.sh
  9. Configure PPTP Network, if desired: cat /root/pptp-faq
  10. Check status to be sure everything is working: status

A Few Important Tips. Every operating system and service provider has their quirks. Ask Bill Gates! Debian and especially Comcast are no different. Fortunately, with Debian, it’s a very short list.

1. Use the following commands (only!) to shutdown and restart your server: halt and reboot. These commands are reworked in Incredible PBX to gracefully shutdown important services so that files don’t get damaged. Please use them!

2. If you ever want to move your server to a different network, complete these steps before you leave your existing network. First, using add-ip or add-fqdn, add the new WhiteList addresses for your new location using Option 0 (all privileges). Otherwise, you won’t be able to access your server once you move. Then issue the commands below. This will trigger a new Phase I update (outlined above) on the default network (eth0) using DHCP the next time you boot your Pogoplug.

touch /etc/update_hostconfig
halt

3. You really do need email connectivity to get the most out of Incredible PBX. It’s the way you receive important notifications from FreePBX, and it’s also how voicemail messages are delivered. From the Linux CLI, test your server to be sure you can send emails reliably:

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

After checking your spam folder, if you really didn’t get the email, it may be that your service provider is blocking downstream SMTP traffic. You can use your provider’s SMTP server as a smarthost to send out mail with Exim4. Just run the following program to reconfigure the Exim mail server: dpkg-reconfigure exim4-config. Choose the SmartHost option and enter your provider’s SMTP address, e.g. smtp.comcast.net or smtp.knology.net. Exim will restart.

4. If you’d like to activate ODBC support for Asterisk including our ODBC sample applications including Speed Dial, here are the steps. Log into your server as root and issue the following commands:

cd /root
wget http://incrediblepbx.com/odbc-pogoplug.tar.gz
tar zxvf odbc-pogoplug.tar.gz
rm odbc-pogoplug.tar.gz
./mysql-sample
./mysql-odbc
./odbc-gen.sh

Now you can try things out by dialing 222 from a phone connected to your server. When prompted for the employee number, enter 12345. Or dial 223 and, when prompted for the AsteriDex Dial Code, enter 263 (the first 3 letters of the American Airlines entry).

5. Want a list of your completed calls without using FreePBX? It’s easy:

mysql -uroot -ppassw0rd -e "SELECT SUBSTRING(calldate,6,11) AS calldate,clid,src,dst,duration from asteriskcdrdb.cdr WHERE disposition='ANSWERED' ORDER BY calldate DESC"

6. There may be situations in which it is desirable to use wireless networking instead of a wired connection with your Pogoplug. For under $10, you now can add WiFi. Here’s our post on the PIAF Forum to show you how.

Managing FreePBX with Incredible Backup and Restore

Unlike other releases of Incredible PBX, the backup and restore tools can be helpful on the Pogoplug platform. Even though Asterisk runs smoothly, calls sound great, and performance is pretty amazing, the FreePBX GUI is usable but a bit sluggish on the Pogoplug platform. If the performance bothers you, there’s a workaround. Create an Incredible Backup image of your Pogoplug, restore that image on a more normal Ubuntu 14 platform with ample RAM, and then make your FreePBX changes there using the FreePBX GUI. When you’re finished, make a backup of the changes, and then restore that backup to your Pogoplug. It sounds more complicated than it actually is. In essence, you’ll be transforming FreePBX into an Asterisk code generator. In fact, once a backup is restored, you can shut down your web server, and almost everything will still work. We were able to perform the entire procedure including updating all of the FreePBX modules and adding a Google Voice trunk in about 15 minutes using a snapshot of an Incredible PBX for Ubuntu 14 droplet we previously had created. Here are the actual steps to perform the first time:

1. Take an image snapshot of your server with Incredible Backup: /root/incrediblebackup

HINT: No need to do it initially. One is included: /backup/DU-2014.09.07.19.46-A11.12.0-F2.11-I11.12.0.tar.gz

2. Create a 512MB Droplet on Digital Ocean using Ubuntu 14 and Incredible PBX for Ubuntu. Follow the Nerd Vittles tutorial which also has a signup link to assist our projects. Coupon code: ALLSSD10 gets you a $10 credit this month. Once you’re up and running, you may want to take a snapshot so that you can quickly recreate droplets while also avoiding hourly charges for the one you’ve previously built (whether running or not!). Digital Ocean 512MB droplets cost less than a penny an hour so this is not a big ticket item. When you finish with the droplet, just destroy it (once you’ve made a snapshot!). Then the money meter stops. First time build takes about 30 minutes.

3. After creating /backup folder on DO droplet, copy your backup image from step #1 to this folder.

4. Restore the image: /root/incrediblerestore /backup/DU-somefilename.tar.gz

5. Open FreePBX on DO with a browser and log in as admin with your admin password.

6. Make all the changes desired using the tutorial below. Reload FreePBX (red bar) when prompted before exiting!

7. Make a DO backup of your new setup: /root/incrediblebackup

8. Copy the DO backup file to /backup on your Pogoplug.

9. Restore the DO backup: /root/incrediblerestore /backup/DU-somefilename.tar.gz

10. Log out and back into your Pogoplug as root.

Getting Started with VoIP and FreePBX

To access FreePBX, just point to the IP address of the server. The main control panel looks like this:

As configured, the default user account for FreePBX administration is admin. The password is whatever you set in the initial setup above. If you ever forget it, you can reset it easily: /root/admin-pw-change.

For those new to Asterisk and FreePBX, here’s a brief primer on what needs to happen before you can make and receive calls. If you have an existing Google Voice account, lucky you. This gets you a phone number for your PBX so people can call you. And it provides a vehicle to place free calls to plain old telephones in the U.S. and Canada so long as Google continues to provide the free service.

If you don’t have a Google Voice account or a shiny new smartphone, then you will need to purchase a SIP trunk from one of the numerous vendors around the world. Our favorite (because they provide terrific service at a modest price AND provide financial support to the Nerd Vittles, PBX in a Flash, and Incredible PBX projects) is Vitelity. Their special rates and a link for a discount are included at the end of today’s article.

Unlike POTS phone service from Ma Bell, the SIP World is a little different. First, you don’t need to put all your eggs in one basket. A trunk that gets you a phone number for incoming calls need not be with the same vendor that provides a trunk to place outbound calls. In fact, you may want multiple trunks for outbound calls just to have some redundancy. A list of our favorites in the U.S. and Canada is available on the PIAF Forum. Of course, there also are providers that offer all-you-can-eat calling plans. Two of our favorites are Vestalink and Future-Nine.

You’ll also need a softphone or SIP phone to actually place and receive calls. YATE makes a free softphone for PCs, Macs, and Linux machines so download your favorite and install it on your desktop.

Phones connect to extensions in FreePBX to work with Incredible PBX. Extensions talk to trunks (like Google Voice) to make and receive calls. FreePBX uses outbound routes to direct outgoing calls from extensions to trunks, and FreePBX uses inbound routes to route incoming calls from trunks to extensions to make your phones ring. In a nutshell, that’s how a PBX works.

There are lots of bells and whistles that you can explore down the road including voicemail, conferencing, IVRs, autoattendants, paging, intercoms, CallerID lookups, announcements, DISA, call parking and pickup, queues, ring groups, and on and on. And then there’s all of the Incredible PBX applications which are covered separately in this Nerd Vittles article. Once you’re comfortable with one server, you or your company will want some more. This Nerd Vittles article will walk you through interconnecting them into a seamless mesh network so that you can call from one office to another transparently. Yes, those articles were written for the Raspberry Pi. But the beauty of Incredible PBX is that it runs (almost) identically on every server platform.

Here’s our 7-Step Checklist to Getting Started with FreePBX:

1. Setting Up Google Voice. If you want free calling in the U.S. and Canada, then you’ll need an existing Google Voice account that includes the Google Chat feature. You’ll need one dedicated to Incredible PBX, or it won’t work. Log out after setting up the new Google Voice account! Also note that Google Voice may cease to function at any time after May 15, 2014. You can read all about it here.

  • Log into existing Google Voice account
  • Enable Google Chat as Phone Destination
  • Configure Google Voice Calls Settings:
    • Call ScreeningOFF
    • Call PresentationOFF
    • Caller ID (In)Display Caller’s Number
    • Caller ID (Out)Don’t Change Anything
    • Do Not DisturbOFF
    • Call Options (Enable Recording)OFF
    • Global Spam FilteringON

  • Place test call in and out using GMail Call Phone
  • Log out of your Google Voice account

If this fails, then Google may have blocked your IP address. Here’s how to unblock it.

2. Activating a Google Voice Trunk. To create a Trunk in FreePBX to handle calls to and from Google Voice, you’ll need three pieces of information from the Google Voice account you set up above: the 10-digit Google Voice phone number, your Google Voice account name, and your Google Voice password. Choose Connectivity -> Google Voice (Motif) from the FreePBX GUI. The following form will appear:

Fill in the blanks with your information and check only the top 2 boxes. If your Google Voice account name ends in @gmail.com, leave that out. Otherwise, include the full email address. Then click Submit Changes and Apply Config.

To activate a Google Voice trunk, you must restart Asterisk on the Pogoplug platform: amportal restart.

3. Setting a Destination for Incoming Calls. Now that you’ve created your Google Voice Trunk, we need to tell FreePBX how to process inbound calls when someone dials your Google Voice number. There are any number of choices. You could simply ring an extension. Or you could ring multiple extensions by first creating a Ring Group which is just a list of extension numbers. Or you could direct incoming calls to an Interactive Voice Response (IVR) system. By default, Incredible PBX is configured to route all incoming calls to extension 701. You can change the setting whenever you like by choosing Connectivity -> Inbound Routes -> Default. In the Set Destination section of the form, change the target destination from the pull-down lists.

Always click Submit and then click Apply Config to save new settings in FreePBX. This is especially important on the Pogoplug platform because you cannot actually do it once you restore the backup image to the Pogoplug.

4. Activating Additional Trunks with FreePBX. As we mentioned, there are lots of SIP providers to choose from. Once you have signed up for service, configuring the trunk is easy. Here is a quick Cheat Sheet courtesy of Kristian Hare, who translated our original setups into a spreadsheet. Just click on the image below to open it in a new window. Then click on the redisplayed image to enlarge it. The left and right cursor keys will move you around in the image. Click on the image again to shrink it.

5. Changing Extension Passwords. From the main FreePBX GUI, choose Applications -> Extensions. Then click on 701 in the Extension List on the right side of your display. You’ll see a form that looks like this:

For now, we only need to make a few changes. First, you need a very secure password for both the extension itself and your voicemail account for this extension. The extension secret needs to be a combination of letters and numbers. The Voicemail Password needs to be all numbers, preferably six or more. Replace the existing password entries with your own (very secure) entries. You also need to lock down this extension so that it is only accessible from devices on your private LAN. You do that with the deny and permit entries which currently are filled with zeroes. Leave the deny entry the way it is which tells Incredible PBX to block everybody except those allowed in the permit entry below. For the permit, we need the first three octets of your private LAN address, e.g. if your LAN is 192.168.0.something then the permit entry will be 192.168.0.0/255.255.255.0.

Finally, you need to plug in your actual email address in the Voicemail section so that voicemails can be delivered to you when someone leaves a message. You can also include a pager email address if you want a text message alert with incoming voicemails. If you want the voicemails to automatically be deleted from the server after they are emailed to you (a good idea considering the disk storage limitations of your microSD card), change the Delete Voicemail option from No to Yes. That’s it. Now save your settings by clicking the Submit button. Then reload the dialplan by clicking on the red prompt when it appears.

In case you’re curious, unless you’ve chosen to automatically delete voicemails after emailing them, you can retrieve your voicemails by dialing *98701 from any extension on your phone system. You’ll be prompted to enter the voicemail password you set up. In addition to managing your voicemails, you’ll also be given the opportunity to either return the call to the number of the person that called or to transfer the voicemail to another extension’s voicemail box. And you can always leave a voicemail for someone by dialing their extension number preceded by an asterisk, e.g. *701 would let someone leave you a voicemail without actually calling you.

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

7. Configuring CallerID Superfecta. In order to match names with phone numbers, Incredible PBX includes a FreePBX application named CallerID Superfecta. Out of the box, Incredible PBX will work fine if you remember to activate CallerID Superfecta whenever you create a new Inbound Route. The CNAM entries also will be displayed in your CDR reports. For those not in the United States, you may prefer to use a lookup source for your numbers other than the ones preconfigured in CallerID Superfecta. You will find all of the available modules on the POSSA GitHub site. Just download the ones desired into /var/www/html/admin/superfecta/sources and then activate the desired sources in Admin -> CID Superfecta -> Default. You can test your results and the performance using the Debug facility that’s built into the module.

If you’re using FreePBX on an Ubuntu server in the Cloud, now is the time to drop down to the Linux command prompt, log in as root, and make a backup: /root/incrediblebackup. Copy the backup from /backup to the same folder on your Pogoplug and restore it: /root/incrediblerestore /backup/DU-somefilename.tar.gz. Then restart Asterisk on your Pogoplug: amportal restart. Finally, log out and back into your Pogoplug to assure that FreePBX will function properly on that platform.

Adding Speech Recognition for Incredible PBX Applications

We used to include Google’s Speech-to-Text service in earlier Incredible PBX builds. Unfortunately, Google has changed the rules a bit. Assuming your server still meets the "personal and development" standard, you can obtain an API key from Google and reactivate speech-to-text functionality for many of the Incredible PBX applications including Weather Reports by City (949), AsteriDex Voice Dialing by Name (411), SMS Dictator (767), and Wolfram Alpha for Asterisk (4747). To activate the STT service, just complete the steps in our tutorial. Then sign up for a Wolfram Alpha App ID (tutorial here), and run the following install scripts:

/root/wolfram/wolframalpha-oneclick.sh
cp /root/pygooglevoice/bin/gvoice /usr/bin
ln -s /usr/bin/gvoice /usr/local/sbin/gvoice
cd /root/pygooglevoice
python setup.py install
/root/smsdictator/sms-dictator.sh

Configuring a YATE Softphone for Pogoplug

As we mentioned, the easiest way to get started with Incredible PBX is to set up a free YATE softphone on your Desktop computer. Versions are available at no cost for Macs, PCs, and Linux machines. Just download the appropriate one and install it from this link. Once installed, it’s a simple matter to plug in your extension 701 credentials and start making calls. Run the application and choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of Incredible PBX on the Pogoplug, 701 for your account name, and whatever password you’re using 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 your first call. It’s that easy!

Introducing Incredible PBX 11.12.0 for the Pogoplug

For those of you that missed last week’s article on the CuBox platform and are new to Asterisk and the world of VoIP telephony, let us take a moment and explain how Incredible PBX fits into the puzzle. For lack of a better term, Incredible PBX is a turnkey aggregation in a bootable image that is based upon a superset of Debian 7 packages plus Asterisk, the FreePBX GUI, and a sizable collection of applications for the Asterisk platform. You download a tarball, decompress it, write the image file to an SD card, insert the card into your Pogoplug 4, and presto! You’ve got a turnkey PBX. Add credentials for a trunk or two to make and receive calls, connect some phones, and your SOHO office or home will come alive with a versatile PBX platform that used to cost organizations hundreds of thousands of dollars. What’s included in Incredible PBX? Glad you asked. Here’s a 3-minute video showcasing a few of our favorite Incredible PBX text-to-speech applications:


The Incredible PBX 11 Inventory. Here’s the current feature set on the Pogoplug platform. In addition to its superset of hundreds of Debian 7 packages, Asterisk 11, and FreePBX 2.11 with the Lighttpd web server, Exim 4 mail server, MySQL, PHP, phpMyAdmin, and the IPtables Linux firewall, check out these additions:

A Few Words About Security. Thanks to its Zero Internet Footprint™ design, Incredible PBX is different. It remains the most secure Asterisk-based PBX around. What this means is Incredible PBX has been engineered to sit anywhere, either behind a NAT-based, hardware firewall or directly on the Internet. No device other than those on your private LAN, a few of the major (trusted) SIP providers around the world, and those that you authorize on your WhiteList can even see your server. Additional IP addresses can be added to the WhiteList by the administrator registering new IP addresses using add-ip or add-fqdn from the Linux CLI. Read about this $100,000 VoIP phone bill, and you’ll better appreciate why WhiteList-based server security has become absolutely essential. WhiteList Security means only those devices with a registered IP address in your WhiteList can get to your server’s resources. To the NSA and everyone else, your server doesn’t even show up on the radar. Their only way to contact you is a POTS telephone using your published phone number. Our complete tutorial on Travelin’ Man 3 is available here. With Incredible PBX for the Pogoplug 4, it’s installed and preconfigured. Enjoy!


Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number.

Originally published: Monday, September 8, 2014


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Our forum is extremely friendly and is supported by literally hundreds of Asterisk gurus. In fact, we already have a thread underway on the Pogoplug adventure.



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

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