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Introducing Incredible PBX with XiVO Snapshots

Monday, October 10, 2016 / 1 Comment on Introducing Incredible PBX with XiVO Snapshots

If you’ve been following along in our XiVO adventure with Incredible PBX, you already know that there were a significant number of configuration hoops to jump through once the base install was finished. While these steps are well documented in the original Incredible PBX for XiVO tutorial, there still were plenty of opportunities for typos and skipping steps. Any misstep could spell the difference in a perfectly functioning PBX and one that couldn’t make or receive calls. Today we’re pleased to report that approach is now going the way of cars with a stick shift. If you want to continue to manually configure your XiVO PBX, you still have that option. Just jump to the original tutorial and run the installer choosing the options you wish to activate. But if you prefer a self-driving Tesla, that’s now an option as well. Continue reading, and we’ll walk you through using XiVO Snapshots.

A XiVO Snapshot is just what the name implies. It’s a snapshot of a working XiVO PBX that has virtually everything already configured: SIP settings to work with Asterisk®, a SIP extension to work with a SIP phone, or softphone, or WebRTC plus your cellphone, SIP and Google Voice trunk setups for most of the major commercial providers, and default inbound and outbound routes to ease the task of routing calls into and out of your PBX. Basically, you plug in your credentials from your favorite provider after running the Incredible PBX for XiVO installer with all Incredible PBX options enabled. Then you tell XiVO how to route the calls, and you’re done. You can have a stable and functional PBX making calls to anywhere in the world in a matter of minutes. Then you can review our numerous tutorials to add additional bells and whistles while you’re already enjoying a fully functional PBX.

Incredible PBX for XiVO Installation Overview

Before we roll up our sleeves and walk you through the installation process, we wanted to provide a quick summary of the 10 Basic Steps in setting up Incredible PBX for XiVO. By the way, the whole process takes less than an hour!

  1. Set Up Desired PBX Platform: Stand-alone PC, Virtual Machine, or Cloud-Based Server
  2. Run the Incredible PBX for XiVO installer and Activate All Options
  3. Set Up One or More SIP or Google Voice Trunks for Your PBX
  4. Tell XiVO Where to Direct Incoming Calls from Each Trunk
  5. Tell XiVO Which Trunk to Use for Every Outbound Calling Digit Sequence
  6. Set Up a SoftPhone or WebRTC Phone (or both)
  7. Decide Whether to Activate Simultaneous Ringing on your Cellphone
  8. Add Google Speech Recognition Key (if desired)
  9. Activating DISA with Incredible PBX for XiVO (if desired)
  10. Test Drive Incredible PBX for XiVO

1. Incredible PBX for XiVO Hardware Platform Setup

The first step is to choose your hardware platform and decide whether you want to babysit a server and network or leave those tasks to others. We’ve taken the guesswork out of the setups documented below. The last four options are cloud providers, each of whom provides a generous discount to let you kick the tires. So click on the links below to review the terms and our walkthrough of the setup process on each platform.

If your situation falls somewhere in between all of these, here’s a quick summary. For stand-alone systems and virtual machine platforms that you own (such as VirtualBox and VMware ESXi), download and install the 64-bit version of XiVO using the XiVO ISO. For most other virtual machine platforms in the Cloud, you’ll start by creating a 64-bit Debian 8 virtual machine with at least 1GB of RAM and a 20GB drive.

2. Running the Incredible PBX for XiVO Installer

Once you have your hardware platform up and running, the rest of the initial setup process is easy. Simply download and run the Incredible PBX for XiVO installer. On some platforms, it first updates Debian 8 to current specs and reboots. Then log back in and rerun the installer a second time. You will be prompted whether to activate about a dozen applications for Incredible PBX. Choose Y for each option if you want to take advantage of the XiVO Snapshot with all components preconfigured. Otherwise, you’ll need to jump over to the original tutorial and manually configure all of the XiVO components.

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-XiVO.sh
chmod +x IncrediblePBX13-XiVO.sh
./IncrediblePBX13-XiVO.sh

3. Setting Up SIP and Google Voice Trunks with XiVO

There are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for XiVO server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, XiVO Snapshots goes a step further and actually creates the trunks for you, minus credentials, as part of the initial Incredible PBX install.

For Google Voice trunks, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.

For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the XiVO web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required when using the XiVO Snapshot.

4. Directing Incoming Calls from XiVO Trunks

Registered XiVO trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the XiVO web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.

5. Routing Outgoing Calls from XiVO to Providers

Outgoing calls from extensions on your XiVO PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell XiVO which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell XiVO to send calls to Vitelity if the caller dialed a 10-digit number. XiVO has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell XiVO to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the XiVO Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:

6. Setting Up Softphone & WebRTC to Connect to XiVO

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of XiVO, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the XiVO Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the XiVO platform, do NOT use an actual extension number for your username with XiVO. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.

WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for XiVO. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the XiVO Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password.

To use WebRTC, you first need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, visit the XiVO WebRTC site. The Username is 701. The Password is the one you obtained above. The IP Address is the address of your XiVO PBX.

7. Setting Up a CellPhone Extension with XiVO

In addition to ringing your SIP extension when incoming calls arrive, XiVO can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then XiVO will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for XiVO comes preconfigured with cellphone support for extension 701. To enable it, access the XiVO Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You can answer the incoming calls on either your cellphone or the phone registered to extension 701.

8. Activating Voice Recognition for XiVO

Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your XiVO PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Don’t forget to add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.

9. 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.

There are three ways to implement DISA with Incredible PBX for XiVO. You can continue reading this section for our custom implementation with two-step authentication. There also are two native XiVO methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.

We prefer 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. To get started, edit /root/disa-xivo.txt. 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

2. 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.

3. 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)

4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.

5. 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 reload

6. 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 is installed, 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)

7. 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)

A sample is included in the XiVO Snapshot. Here’s how ours looks for the Nerd Vittles XiVO Demo IVR:



8. 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.

10. Test Drive Incredible PBX for XiVO

To give you a good idea of what to expect with Incredible PBX for XiVO, we’ve set up a sample IVR using voice prompts from Allison. Give it a call and try out some of the features including voice recognition. Dial 1-843-606-0555.

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)

What To Do and Where to Go Next?

Here are a Baker’s Dozen projects to get you started exploring XiVO on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Note that all of these components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with XiVO. There’s a treasure trove of information that awaits.

XiVO and Incredible PBX Dial Code Cheat Sheets

Complete XiVO documentation is available here. But here are two cheat sheets in PDF format for XiVO Star Codes and Incredible PBX Dial Codes.

Published: Monday, October 10, 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…

Integrating SIP URIs into XiVO for Free Worldwide Calling

Monday, September 26, 2016

It’s been a while since we’ve explored SIP URIs and all of the advantages that SIP URI calling brings to your PBX. Number one on that list is FREE calling to and from anyone on the planet so long as both of you have an Internet connection with a SIP phone or a VoIP server such as Incredible PBX for XiVO. SIP URIs are the fundamental building blocks for VoIP technology. Consider this. If everyone in the world had a SIP address instead of a phone number, every call to every person in the world via the Internet would be free. That pretty much sums up why SIP URIs are important. The syntax for SIP URIs depends upon your platform. With Asterisk® they look like this: SIP/somebody@FQDN.yourdomain.com. On SIP phones, SIP URIs look like this: sip:somenameORnumber@FQDN.yourdomain.com. Others use somenameORnumber@FQDN.yourdomain.com. Assuming you have a reliable Internet connection, once you have “dialed” a SIP URI, the destination SIP device will ring just as if the called party had a POTS phone. Asterisk® processes SIP URIs in much the same way as other calls originating from trunks and, as noted, SIP URI calls of any duration to anywhere are free. Today we’ll show you how to set things up on your XiVO PBX without exposing any ports to the Internet in a way that would jeopardize your server’s security.

Placing Outbound SIP URI Calls with a SIP Softphone

There are two ways to place outbound SIP calls. You can use a SIP phone or softphone that supports SIP URI calling to dial SIP URIs directly. If you have a Mac, the best free softphone for SIP URI calling is Telephone which you can download from the App Store. On other platforms as well as Macs, Zoiper is a great no-cost option. Both of these softphones support the sip:someone@FQDN.yourdomain.com syntax. An excellent way to test this is to call our friend Lenny and strike up a conversation: sip:2233435945@sip2sip.info.

Configuring Outbound SIP URIs with XiVO

The major drawback of SIP URIs is they’re difficult both to remember and to dial. It’s much simpler to dial a short number using a traditional phone. And, with Incredible PBX for XiVO, it’s easy to create custom extensions that can be accessed simply by dialing a few digits from any phone connected to your server. Here’s how to set it up in the XiVO GUI.

1. Create a User and assign the Customized Protocol and an Extension Number to that user:

TIP: If you’d prefer to use a different series of numbers for speeddials so you don’t get them mixed up with your standard extension numbers, just add a new range of numbers for XiVO: IPX Configuration → Contexts → Default → Users. Then choose one of them above.

2. Access the new Line that was generated for the new User:

3. Replace the Interface entry for the Line with the desired SIP URI for your speeddial, e.g. SIP/2233435945@sip2sip.info. Then SAVE your new Line settings.

4. Dial 750 from an Extension on your XiVO PBX to try out Lenny using your new SIP URI.

A Better Way to Create SpeedDials with XiVO

We’ve gone through the XiVO GUI approach to demonstrate that it is indeed possible to create speeddials for SIP URIs. However, there is a better way unless you’re one of the naysayers that believes everything is better in a GUI. If you have dozens or even hundreds of speeddials to create, you may change your mind. The GUI approach could obviously become tedious. Instead, with one line of Asterisk dialplan code, you can create as many speeddials as you like keeping in mind that it’s your responsibility to assure that SIP URI extension numbers don’t conflict with existing extensions on your server. Insert a new section of code at the bottom of /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf and reload your dialplan: asterisk -rx "dialplan reload".

You can also insert this code from within the XiVO GUI itself: IPX Configuration → Configuration Files. Edit xivo-extrafeatures.conf and insert the following code snippet at the end of the file and Save your entries. The dialplan will be reloaded automatically.

Some of our favorites include the following:

;# // BEGIN SpeedDials
exten = 882,1,Dial(SIP/200901@login.zipdx.com)     ; V-U-C on Fridays at noon EST
exten = 8378,1,Dial(SIP/thetestcall@getonsip.com)  ; T-E-S-T everything VoIP
exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)  ; L-E-N-N-Y
exten = 68742,1,Dial(SIP/0289304@zero-nine.biz)    ; M-U-S-I-C
exten = 3733411,1,Dial(SIP/411@ideasip.com)        ; F-R-E-E-4-1-1 Directory Asst
;# // END SpeedDials

Creating a SIP URI Address for Your XiVO PBX

Free calls to other folks is only half of the story, of course. You’re also going to want a way for people to call you without incurring charges for the calls. There are many SIP URI approaches for inbound calls. Most of them are not safe with Asterisk. Let me say that again. Most of them are not safe with Asterisk. The reason is because most of them force you to open SIP access to your server for everybody in the world. Unfortunately, that means they can not only call you, but they can also attempt to use your extensions and trunks to place very expensive calls to others. Don’t even think about opening the SIP floodgate by exposing port 5060 unless Bill Gates sends you a check every week. You’ve been warned!

Setting Up an iNum SIP URI Trunk with XiVO

The better and safer way to add SIP URI connectivity to your XiVO server is to first obtain a freely available iNum DID from one of the many providers that support iNum and then use that provider as a SIP intermediary. All SIP calls pass only over your registered trunk with your provider. Our favorites in no particular order are VoIP.ms, LocalPhone and CallCentric. There are many, many others. In order to obtain a free iNum DID, you will need an account with one of these providers. All require some sort of minimal deposit, but you usually can get back unused funds if you decide to close your account down the road. Our XiVO tutorials for VoIP.ms, LocalPhone, and CallCentric will walk you through creating your SIP account and registering it with your XiVO server. Then verify that your SIP account is registered:

asterisk -rx "sip show registry"

Configuring an iNum DID with VoIP.ms

Our trunk tutorials for LocalPhone and CallCentric will walk you through their setup procedures for iNUM DIDs. VoIP.ms provides more flexibility in redirecting trunks so let us quickly walk you through their procedure. Log in to your VoIP.ms account and then order your free iNum DID at this link. Your iNum DID then will appear in your DID Listing here. Write down your iNum DID which you’ll need in a minute to configure the XiVO side of things. Then click on the Edit DID icon beside your iNum DID and assign the DID to your registered Main Account or the SubAccount that you’ve already registered with XiVO. Be sure to use the same DID POP that you used when you registered your VoIP.ms account with XiVO. Don’t enable VoiceMail and set the ring time to 60 seconds just to keep things simple.

Configuring XiVO to Support Your iNum DID

Now for the XiVO part. Using a browser, log into the XiVO GUI. Navigate to IPX Configuration → Contexts → Default → Users. For VoIP.ms and LocalPhone, add a new Number Range starting and ending with your iNum DID. Then click Save. For CallCentric, do the same thing but substitute your CallCentric username which will be an 11-digit number starting with 1777.

Repeat the above in IPX Configuration → Contexts → from-extern (Incalls) → Users.

For CallCentric only, also click on the Incoming Calls tab and add a new Number Range. For the Starting value, use your 11-digit LocalPhone username. For the DID length, set it to 11. You do NOT need to include a Number Range ending value. Click Save when you’re finished.

For VoIP.ms, navigate to IPX Settings → Users. Then Add a new User for your iNum DID. In the General tab, name the User VoIP.ms iNum. In the Lines tab, provide your actual iNum DID number. This must be the same number you added to the Number Range in the Default context above. In the No Answer tab, set the Fail option to the Destination of your choice, e.g. an extension, a ring group, an IVR, etc. Then click Save.

For LocalPhone, navigate to Call Management → Incoming Calls and Add a new Inbound Route for your DID specifying the destination for the calls using your iNum DID number:

For CallCentric, navigate to Call Management → Incoming Calls and Add a new Inbound Route using your 11-digit CallCentric username as the DID. Then specify the destination for the calls and click Save.

Calling Your XiVO PBX Using Your iNum SIP URI

To receive SIP URI calls safely on your iNum DID, your SIP URI is your iNum DID number followed by @sip.inum.net, e.g. 883510012345678@sip.inum.net. Neither the identity of your XiVO PBX or your SIP service provider is ever exposed. Enjoy your safe, free calling!

Originally published: Monday, September 26, 2016



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

 

Special Thanks to Our Generous Sponsors


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

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

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

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

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

Some Recent Nerd Vittles Articles of Interest…

2016, The Year of VoIP Choice: Redundancy and Multi-Tenant with Wazo

Monday, September 5, 2016



As we celebrate Labor Day, it seemed appropriate to document why Wazo separates the men from the boys so your phones don’t end up as boat anchors buried in the sand. Today our focus is "High Availability (HA)" and "Multi-Tenant (MT)", two very expensive options for many PBXs including some that loosely tout their platforms as free.

In the PBX context, HA means that, when your server fails, there’s another one waiting in the wings to automatically take over. Much of this technology is based upon open source tools, but Sangoma sells a pair of limited term licenses as a FreePBX® add-on for a cool $3,000 not including hardware AND annual maintenance fees. With Wazo, it’s FREE! You can pair two Raspberry Pi’s or two Cloud servers, or you can mix-and-match with any combination of servers you choose. Here’s how we did it in 3 minutes flat:



Multi-tenant has been discussed for the FreePBX platform for the better part of a decade. As best we can tell, it’s still a pipe dream. Virtual machines running separate servers are the suggested solution even though this requires managing multiple Asterisk platforms forever. With Wazo’s FREE Entities module, MT is a cake walk. We’ll walk you through the 5-minute setup process thanks to the tips provided by Amy Grant on the PIAF Forum.

Deploying Wazo HA Servers with NeoRouter

Here’s the HA setup drill. First, you build two identical Wazo platforms running the same version of Incredible PBX for Wazo. Then you set the first server up as the Master and the second one as the Slave. As we said, these servers don’t need to be on the same hardware platform. And they need not be colocated although they have to share the same private LAN. We’ll handle that little detail by taking advantage of the NeoRouter client software that’s already installed as part of every Incredible PBX for Wazo build.

Unless both of your servers reside on the same local area network, you will need to deploy a NeoRouter server somewhere, but NOT on your Wazo Master since the NeoRouter server itself would become a single point of failure should it die along with your primary server. The Slave server would be a great choice. We covered the NeoRouter Server setup a long time ago in this tutorial, but don’t use the vintage install script. Instead you’ll need to deploy a current version of the Free NeoRouter Server that matches your server platform now that we support operating systems other than CentOS. Incidentally, all of the supported Cloud platforms that we’ve documented for Wazo also support NeoRouter.

We’ve made NeoRouter Server setup easy with this script which works with CentOS/SL, Ubuntu, Debian, and Raspbian. The actual setup steps covered in our original tutorial still are the same.

cd /root
wget http://incrediblepbx.com/install-neorouter-server
chmod +x install-neorouter-server
./install-neorouter-server

After you have your Free NeoRouter Server in place, the next step is to run nrclientcmd on each Wazo server and login to your NeoRouter Server with your credentials. The NeoRouter Server will assign a private IP address to each machine on the NeoRouter VPN. The addresses will be in the range 10.0.0.1 to 10.0.0.255. We’ll use these assigned addresses when setting up the Master and Slave Wazo HA servers.

High Availability Prerequisites with Wazo

In the Incredible PBX for Wazo context, the prerequisites list for your two HA servers is a short one. (1) You need two functioning Incredible PBX for Wazo machines on the same local area network. (2) Both the Master and Slave must be running the same version of Wazo. (3) All trunk registration timeouts (expiry) must be less than 300 seconds. (4) The Slave server must have no phone provisioning plugins installed.

For those using Google Voice trunks with OAuth in conjunction with Incredible PBX for Wazo, keep in mind that this is NOT an integral component of Wazo so it technically is not supported. However, you can easily make it work by configuring any desired Google Voice trunks on BOTH the Master and Slave machines using add-gvtrunk before enabling High Availability. Then the Google Voice trunks will continue to work even after a failover to Slave.

High Availability Limitations with Wazo

When the Master node fails, some features are not available on the Slave:

  • Call history / call records are not recorded.
  • Voicemail messages saved on the Master node are not available.
  • Custom voicemail greetings recorded on the Master node are not available.
  • Phone provisioning is disabled, i.e. a phone will always keep the same configuration, even after restarting it.
  • Phone remote directory is not accessible because the provisioned IP address points to the Master.

Configuring Your Servers for High Availability

Like most Wazo tasks, setting up High Availability on your Master and Slave servers is a 5-minute process. Begin by configuring HA in the Web interface: Configuration ‣ Management ‣ High Availability. (1) Configure the first server as Master with the Remote Address of the Slave. (2) Login to the Linux CLI of Master as root and restart Wazo: xivo-service restart. (3) For the second machine, configure the server as Slave with the Remote Address of the Master.

Next, return to the Linux CLI of Master while still logged in as root. (1) Set up file synchronization by running this script: xivo-sync -i. (2) Start configuration synchronization by running: xivo-master-slave-db-replication 192.168.1.2 using the actual IP address of your Slave. (3) Finally, synchronize the two servers by running xivo-sync on Master. Done! Isn’t it nice saving $3,000 for 5 minutes work using open source software? 🙂

If you love the nitty gritty details, you can read up on Wazo HA in their excellent documentation.

Here’s what pbxstatus will show on Master and Slave while both servers are operational:

And here’s what happens when you halt Master. Within a minute or two, your designated Slave server will come to life:

Choosing Compatible Phones for High Availability

That’s only half the story, of course. Now that you have HA up and running, the remaining trick is that you want your phones to continue to work when things switch over to Slave. To accomplish this, you’ll need to use SIP phones that are compatible with HA technology. Some are, and many are not. Wazo has made it easy for you by publishing a compatibility list. Their documentation includes Officially Supported Devices as well as Community Supported Devices. HINT: Snom, Yealink, and Aastra 6700i and 9000i series phones are your safest bets.1 Here’s what a SIP extension setup would look like on Yealink’s popular T46G:

Deploying Multi-Tenant Technology with Wazo

If you’re new to MT technology, the idea here is to provide separate extensions and trunks for use by different departments within an organization. The reasons should be obvious. These departments have separate budgets and separate clientele, and you probably don’t want the public calling a central number in order to reach everyone in an organization. And the organization wants to identify costs and log calls associated with its various departments.

Wazo handles MT using Entities. When you set up Incredible PBX for Wazo, it automatically created a single Entity named Incredible PBX. You can create additional ones and name them anything you like in the Wazo Web interface: Configuration ‣ Management ‣ Entities.

Next, create Contexts to support your new Entity. Mimic the existing contexts in IPX ‣ IPX Configuration ‣ Contexts and provide unique names for each of them. Be sure you associate each of the new contexts with the new entity you created. Then set up users, lines, trunks, and call routing for the new entity in the same way you did it for the original IncrediblePBX entity. Take a look at Amy Grant’s setup with Google Voice on the PIAF Forum for additional tips. Simple and it’s FREE!

Originally published: Monday, September 5, 2016  Updated: Saturday, January 28, 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. HA failover even works great using $29 UTP-E62 if you can find one. [↩]

Raspberry Pi One-Minute Wonder: A Turnkey and Truly Incredible PBX for XiVO

Monday, August 29, 2016 / 1 Comment on Raspberry Pi One-Minute Wonder: A Turnkey and Truly Incredible PBX for XiVO


Hard to believe it’s been 4½ years since the introduction of the original Raspberry Pi®. We love half-birthdays, and we’ve got a blockbuster gift for you today as we celebrate the fact that almost 10 million RasPi’s have been shipped. Yes, our love affair with the Raspberry Pi lives on. The sensational Raspberry Pi 3 sports a 1.2GHz 64-bit quad-core ARM Cortex-A53 CPU with ten times the performance of the original Raspberry Pi. Of particular interest to the VoIP community will be the RasPi 3’s integrated 802.11n wireless LAN and Bluetooth 4.1 hardware. And, of course, the RasPi 3 retains its compatibility with the Raspberry Pi 1 and 2. Did we mention it’s still just $35? Because we like to celebrate half birthdays, too, we’re pleased to introduce a brand new Incredible PBX™ for XiVO image for the Raspberry Pi 3 featuring Raspbian 8, the latest release of Asterisk® 13, and XiVO. This one installs in under a minute. And, yes, it’s still FREE with pure open source GPL code.

Special Thanks. First things first. We want to extend our extra special thanks to Iris-Network for their awesome Raspivo – XiVO build. Without it and their repositories, none of this would have been possible.

Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful at the $10 price point.1

Raspberry Pi 3 Shopping List. Before you can install Incredible PBX for XiVO, you’ll need a compatible Raspberry Pi 3 platform. Here’s the short list that, when coupled with the Incredible PBX image, turns today’s adventure into kid’s play:

  • $35* Raspberry Pi 3 from MCM or Newark or Amazon
  • $10 Power Adapter (2.5 amps minimum!)
  • $10 32GB microSDHC Class 10 card (Don’t use SanDisk Ultra!)
  • £12.95 Pibow 3 case or $7.50 Official RasPi 3 case

    • About That Asterisk. We write about Asterisk® regularly, but the asterisk we’re talking about is the one accompanying the $35* price tag for the Raspberry Pi 3. Yes, that’s the advertised price. And, no, if you want one quickly, you may pay a bit more. Right now you can snag one on Amazon for $35.99 with two-day Prime shipping. We’re assuming you already own a USB keyboard and an HDMI-compatible monitor. If so, today’s going rate for all of our recommended pieces is under $65, not bad for a fully-equipped, quad-core computer. Did we mention that Incredible PBX for XiVO is FREE with NoGotchas!

    Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, SMS, FAX, SRTP, and OAuth functionality. Voice Recognition and Text-to-Speech VoIP application support using Festival and Google. Free calling with Google Voice, Simonics SIP gateway, or RingPlus cellular service. And all of your Nerd Vittles favorites: AsteriDex, Click-to-Dial, News, Weather, Reminders, and even an Alarm Clock. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Simultaneous Ringing on your Smartphone, Email Delivery of Voicemail, Voicemail Blasting, Automatic Backups, High Availability Support, Automatic Phone Setups, and much more…

    Incredible PBX Network Security Model. Most phone calls cost money. Unlike many of the other "free" VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, it isn’t free at all. Once you plug in that network cable, you’ve painted a bullseye on your checkbook.

    No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in your web server, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. You can’t hack what you can’t see, and the Incredible PBX automatically configures a WhiteList as part of the one-minute setup. And it’s all open source GPL code that you can share with anybody and everybody unlike the so-called "freeware" products. Freeware with Asterisks is anything but free!

    Do your part and do your homework. Comparison shop as if your phone bill matters! 😉 Incredible PBX provides:

    1. Preconfigured IPtables Linux Firewall
    2. Preconfigured Travelin’ Man 3 WhiteLists
    3. Randomized Port Knocker for Remote Access
    4. Fail2Ban Log Monitoring for SSH, Apache, Asterisk
    5. Randomized Ultra-Secure Passwords
    6. Automatic Update Utility for Security & Bug Fixes
    7. Asterisk Manager Lockdown to localhost
    8. Security Alerts via the PIAF Forum

    Incredible PBX for XiVO Installation & Setup Tutorial

    Here’s everything need to know about installation and setup of Incredible PBX for XiVO. "Automatic" means you just watch.

    1. Download and unzip Incredible PBX for XiVO image from SourceForge (includes GV OAuth support)
    2. Transfer Incredible PBX image to microSD card
    3. Boot Raspberry Pi from new microSD card
    4. Login to RasPi console as root:password to initialize your server (Automatic) and expand image to match SD card
    5. Reboot after writing down your server IP address (Automatic)
    6. Login via SSH as root:password to set up passwords (You Pick ’em) & configure firewall (Automatic)
    7. Enjoy!

    Running Incredible PBX for XiVO on the Raspberry Pi

    The standard XiVO boot procedure will begin once you insert your microSD card into the Raspberry Pi 3 and apply power. Within a short time, you’ll get the familiar Linux login prompt. Login as root with a password of password.

    Once you log in, a startup script will briefly configure a few things and then advise you that it’s time to reboot. Write down the IP address provided because for Phase 2 of the setup, we need to use SSH or Putty on the desktop that you will actually be using to manage your server. The reason for this is that Incredible PBX automatically creates a whitelist of IP addresses that the firewall will allow to access your server. If the IP address isn’t in your whitelist, you may lock yourself out except from the RasPi’s console window.

    Once the console window shows that your server has rebooted by displaying the Linux login prompt, switch to SSH or Putty and login as root using the IP address you wrote down. You’ll then be prompted to change your root password for Linux as well as your root password for XiVO GUI access using a web browser. You’ll also need to set a PIN that will be used to authorize access to extension 123 to schedule Telephone Reminders on your server. This completes the configuration. You’ll get a final screen showing the credentials for the preconfigured extension 701 as well as a reminder that your PortKnocker credentials are stored in /root/knock.FAQ in the event you ever lock yourself out of your machine. It’s a good idea to leave this screen displayed while you install and configure a softphone since you can cut-and-paste your extension 701 credentials without having to type anything.

    Once you complete the SIP softphone setup below, you can return to the SSH window and press ENTER to finish the install. The Incredible PBX Automatic Update Utility will run, and then you will be presented with the pbxstatus display. You can access the Asterisk CLI by typing: asterisk -rvvvvvvvvvv. Exit from the CLI by typing quit. As mentioned previously, always shut down your server gracefully by typing halt. When prompted for the hostname, type xivo. Once the shutdown procedure finishes, it’s safe to disconnect the power cord from your Raspberry Pi.

    Beginning with the September 1 release, many of the log files have been disabled to help prolong the life of microSD cards since XiVO tends to be very chatty. If you are running an earlier release, you can follow this tutorial to disable most logging on your Raspberry Pi.

    Enabling WiFi on the Raspberry Pi 3

    With the Raspberry Pi 3, wi-fi hardware is included. The next step is configuring it to connect to your WiFi router. Simply open /etc/wpa_supplicant/wpa_supplicant.conf with nano and (1) edit the SSID name and password fields to authorize access to your local, password-protected WiFi router as well as any open WiFi network. (2) Also update the country code for your WiFi region, e.g. country=US. Then (3) save your changes: Ctrl-X, Y, then press ENTER.

    network={
 ssid="YourSSID"
 psk="YourSSIDpassword"
 key_mgmt=WPA-PSK
 scan_ssid=1
 priority=5
}

network={
 key_mgmt=NONE
 priority=1
}

    Next, enable automatic startup of the wlan0 network interface:

    sed -i 's|#allow-hotplug wlan0|allow-hotplug wlan0|' /etc/network/interfaces

    Finally, stop and restart the wlan0 interface, count to 15, and check pbxstatus to decipher the added private IP address for your WiFi connection:

    ifdown wlan0
ifup wlan0
pbxstatus

    If you want to run your Raspberry Pi exclusively off the WiFi connection going forward, simply unplug the network cable from your RasPi and reboot your server.

    Choosing a SIP Softphone for Incredible PBX for XiVO

    Softphones tend to be a matter of taste for most folks so we’ll keep our suggestions to a minimum. On the Windows platform, it’s hard to go wrong with X-Lite. It works out of the box by simply plugging in the IP address of your server and your SIP username and password. It also happens to be free. The only downside is that X-Lite has a nasty habit of embedding time bombs in their free software so you may have to reinstall it from time to time. If you know what you’re doing Zoiper is another alternative but be advised that it doesn’t work out of the box on servers behind NAT-based routers.

    On the Mac platform, our favorite free softphone is Telephone. It’s a barebones SIP client that just works. As with X-Lite, you plug in your server’s IP address and SIP credentials, and you’re in business.

    On the Linux or Solaris platforms, we assume that you know what you’re doing and that you are perfectly capable of choosing and installing a SIP phone that meets your requirements.

    Incredible PBX Application Quick Start Guide

    We’ve finished the basic Incredible PBX for XiVO setup. You now have a functioning PBX with dozens of applications for Asterisk that work out of the box. It’s probably a good idea to spend a little time getting acquainted with Incredible PBX for XiVO before you add trunks to communicate with the outside world.

    Here’s a handy cheat sheet for some of the Incredible PBX applications that have been installed or are available as add-ons. There’s also a link for more information. This remains a work-in-progress so expect more applications in coming weeks.

    How To Make Easily Compressed Backups of Incredible PBX

    MicroSD cards WILL wear out especially on XiVO servers with lots of activity. So it’s important to make regular backups of your media so you don’t get surprised when things come unglued down the road. After considerable discussion on the PIAF Forum, here’s the collective wisdom.

    You’ll need another machine (such as a Mac or Linux box) on which to plug in the microSD card in order to make a backup image of it since you can’t back up a card that is actually providing the live platform for your PBX. The recommended methodology goes like this. Before shutting down your PBX and removing the microSD card to make the backup, convert all of the unused space on the card to zeros so that the unused space can be easily compressed when you create the backup image. You do this by issuing the following command after logging into the Linux CLI as root on your RasPi 3. Be sure to do it during a period of inactivity on your PBX as it is processor intensive. Then halt the machine and remove the microSD card.

    xivo-service stop
cat /dev/zero > wipe.it ; rm wipe.it
halt

    Insert the card into an SD card slot on the machine you will use to make the backup image and issue the following commands after deciphering the correct device name for your card (/dev/disk4 in this example) using the df utility:

    sudo df -h
sudo dd bs=1m if=/dev/disk4 | gzip -c > incrediblepbx-xivo.img.gz
sudo sync
sudo diskutil eject /dev/disk4s1
echo "It's safe to remove the microSD card now."

    Now return the microSD card to your Raspberry Pi 3 and boot. Store your backups in a safe place!

    Configuring Trunks and Routes with Incredible PBX for XiVO

    The next step in your XiVO adventure is connecting your PBX to the outside world so that you can make and receive phone calls from anywhere in the world. For this you’ll need one or more trunks. Unlike the Ma Bell world, there’s no reason to put all your eggs in one basket. You can use one or more trunk providers for incoming calls with separate phone numbers for each. And you can use one or more trunk providers for outgoing calls and save money on calls to certain countries by choosing the best provider for where you want to call. And, of course, if you live in the United States, you can set up one or more Google Voice trunks and make calls to the U.S. and Canada for free. We’ve written a number of tutorials to make it easy to set up these trunks.

    To get started, point a web browser to the IP address of your PBX. Login as root with the XiVO GUI password you set up above. If you ever forget your password, you can run /root/admin-pw-change to reconfigure it.

    XIVO Trunk Implementation Tutorials

    Once you’ve added one or more trunks, you’ll need to tell XiVO how to route outgoing and incoming calls. Here are our step-by-step tutorials on setting up Outbound Calling Routes and Incoming Call Routes:

    XIVO Call Routing Tutorials

    Enabling Bluetooth & Proximity Detection on the Raspberry Pi


    Where To Go Next with Incredible PBX for XiVO

    Now you’re ready to explore. We recommend you pick up here in our Incredible PBX for XiVO tutorial. And be sure to check out the Last Minute Fixes that didn’t make it into the current build. Enjoy the ride!

    Originally published: Monday, August 29, 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…

    1. Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. [↩]
    2. Vitelity is a platinum sponsor of Nerd Vittles, and they also happen to be the best in the business. You’ll find a discount coupon to get a great deal on a DID and 4-channel trunk toward the end of this article. [↩]

    2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business

    Thursday, May 5, 2016 / 1 Comment on 2016, The Year of (real) VoIP Choice: Meet XiVO, a UC Solution for Any Business

    We promised you that 2016 was going to be a year filled with surprises, and today we’re pleased to introduce another open source, pure GPL3 solution for any business. Whether your requirements are a call center or a versatile phone system for hundreds of employees, XiVO™ offers a compelling unified communications solution that checks all the boxes. Unlike some products that function merely as a code generator for Asterisk®, XiVO is in a league of its own. XiVO is actually an integral component of the Asterisk application itself. It manages your telephony server in realtime using its versatile PostGreSQL database platform. Did we mention it’s also a great playground for hobbyists and SOHO VoIP enthusiasts? Let’s get started.

    UPDATE: The first release of Incredible PBX for XiVO is now available here. Please consider this article superseded by the new release.

    There’s no way to do justice to a product like XiVO in a single article. So our plan is to introduce XiVO today and get your platform up and running where you can make and receive free calls throughout the United States and Canada. Then you can add Incredible PBX components and additional SIP providers as we continue to build them out. Just follow along with our Incredible PBX development for XiVO on the PIAF Forum, and you’ll get a first-hand look at how sausage is made. We already have text-to-speech applications for news and weather up and running. You can take them for a test drive by calling the XiVO demo:

    And, of course, we’ve integrated the Travelin’ Man 3 IPtables firewall to provide rock-solid security for XiVO, and we’ll cover that today as well. As part of this development process, you’ll discover how easy it is to build Asterisk applications for XiVO on your own. And hopefully you’ll share some of your creations with the rest of us. That’s what open source development is all about.

    Choosing an Experimental Platform for XiVO

    We’re just getting started with XiVO development so, like us, we’re assuming you’ll want to kick the tires a bit before jumping into a new VoIP solution for the long haul. That means you first must choose a platform on which to install XiVO. We have several recommendations for you. If you have a robust desktop machine with lots of RAM and processing power, then installing XiVO under VirtualBox may be the way to go. We actually use an iMac with 16GB of RAM, and it provides plenty of horsepower to run VirtualBox and XiVO. With VirtualBox, we’ll start by downloading the XiVO ISO.

    We didn’t mention that XiVO has been under development for over 10 years and is supported by the original developers with financial support from Avencall. Because of its Canadian roots, it seems only fitting that many may wish to consider CloudAtCost in Canada as an appropriate site to host your experimental XiVO server. A one-time payment of $10.50 still buys you a sandbox in the cloud for life with coupon code TAKE70, and XiVO installs on the CloudAtCost platform without a hiccup. For a CloudAtCost implementation, we’ll start by creating a Debian 8 server.1 And then we’ll download and run the XiVO installation script to build our XiVO server. Finally, we’ll walk you through setting up XiVO on a $5/month Digital Ocean Droplet which provides state-of-the-art performance at rock-bottom Cloud pricing. So begin by choosing your hardware platform from the three options below:

    1. Installing XiVO as a VirtualBox Virtual Machine

    For standalone implementations including VirtualBox, we’ll begin by downloading the 64-bit XiVO Server ISO to your desktop. Next, create a VirtualBox 64-bit Debian VM platform with 1024 MB RAM and at least a 10GB virtual drive. In System Settings, enable I/O APIC and disable the other options. Select a Sound Card to match your machine and configure Network Adapter 1 as a Bridged Network Device. In the Storage Settings (shown below) for your (1) Empty IDE Controller, (2) select the downloaded XiVO ISO as your installation media. Start the VM and proceed through the initial install.

    Click Install, choose your language, pick your time zone, choose your keyboard map, create a very secure root password, and choose a Debian mirror that’s close to your server. Choose /dev/sda as your bootloader assuming that’s the disk drive configured by VirtualBox. In less than 10 minutes, the install will complete and your VM will reboot. Log into your server as root and obtain your IP address: ifconfig. You’ll need it for the web configuration step that comes next.

    2. Installing XiVO as a CloudAtCost Cloud-Based Server

    You can’t use an ISO as the installation media at CloudAtCost so we have to start by building a 64-bit Debian 8 virtual machine with at least 512 MB RAM and a 10GB virtual drive. No need to choose a larger drive at the moment since there’s a bug in CloudAtCost’s installer for Debian 8. See the footnote for details. Once your virtual machine is built, log in as root and issue the following commands to kick off the XiVO install:

    apt-get -y remove apache2*
apt-get update
apt-get -y upgrade
reboot
# log back in as root and...
wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh
bash xivo_install_current.sh

    3. Installing XiVO as a Digital Ocean Droplet

    As with CloudAtCost, you’ll need to begin your XiVO adventure at Digital Ocean by first signing up for an account. With our referral code, you’ll get a $10 credit (and so will Nerd Vittles). That’s good for two full months of service to kick the tires of XiVO without ever spending a dime. Once your account is set up, create a $5/month Debian 8 (64-bit) Droplet. When you receive the email with your droplet credentials, log into your new server as root using SSH/Putty and issue the following commands to get Debian 8 squared away:

    apt-get update
apt-get upgrade -y
dd if=/dev/zero of=/swapfile bs=1024 count=1024k
chown root:root /swapfile
chmod 0600 /swapfile
mkswap /swapfile
swapon /swapfile
echo "/swapfile          swap            swap    defaults        0 0" >> /etc/fstab
sysctl vm.swappiness=10
echo vm.swappiness=10 >> /etc/sysctl.conf
free
reboot

    After the reboot, log into your server again with your new root password and kick off the XiVO install:

    wget http://mirror.xivo.io/fai/xivo-migration/xivo_install_current.sh
bash xivo_install_current.sh

    Enabling SSH Root Access with XiVO

    If you installed XiVO using the XiVO ISO, then root logins via SSH are disabled by default. Only enable it if you plan to also implement the firewall in the next step! To enable root logins via SSH, log into the server console as root and edit the SSH config file: nano -w /etc/ssh/sshd_config. Find the line in the Authentication section that begins with PermitRootLogin and change it to: PermitRootLogin yes. Save your change (Ctrl-X, y, ENTER) and then restart SSH: /etc/init.d/ssh restart.

    Setting Up a Firewall to Protect XiVO

    We don’t build PBXs without a rock-solid firewall, but it’s your phone bill so the choice is all yours. The Travelin’ Man 3 implementation of the Linux IPtables firewall provides a safe computing platform using a WhiteList to only allow access by trusted users and providers. You can add additional users to the whitelist as desired using add-ip and add-fqdn in the /root folder. Restart your firewall using only this command: iptables-restart. If you’ll be using FQDNs in your WhiteList, then add the ipchecker script to your cron jobs. Then review Step #5 in the TM3 tutorial.

    echo "*/10 5-22 * * * root /root/ipchecker > /dev/null 2>&1" >> /etc/crontab

    It’s imperative that you set this up from a client workstation that’s running SSH or Putty. Otherwise, you may inadvertently lock yourself out from your own server. While logged into your server via SSH as root, issue the following commands:

    cd /root
wget http://incrediblepbx.com/firewall-xivo.tar.gz
tar zxvf firewall-xivo.tar.gz
rm -f firewall-xivo.tar.gz
./tm3-xivo.sh

    Configuring XiVO with a Web Browser

    Once the basic install is completed, you use a web browser to actually configure and manage your XiVO server. To get things started, point your browser to the IP address of your XiVO server. Choose your Language. Accept the GPL3 license agreement. Then fill in the blanks to create a Hostname for your server (XiVO), a domain name (some domain that you own or one chosen from your favorite dynamic DNS provider), a very secure Web interface password (choose as if your phone bill depends upon it). The network interface and DNS server entries should already be correct. Click Next.

    On the second configuration screen, choose an Entity (department/organization name or IncrediblePBX will suffice). Then set up the Contexts to manage calls on your PBX:

    • Internal Calls Context: manages extension numbers that can be reached internally
    • Incalls Context: manages calls coming from outside of your system
    • Outcalls Context: manages calls going from your system to the outside

    Here’s what we’ll be using by way of example:

    Finally, validate your entries to complete the configuration. Now log into your XiVO server as root using your newly created web password. You should get a status screen that looks something like this. If you had any doubts about the quality of the XiVO product, this should put your mind at ease. 🙂

    Logging Into the XiVO Web Interface

    To make changes in your XiVO setup, you’ll need to log into the web interface at the IP address of your XiVO PBX. Login with root as the username together with the Web Interface Password you set up above. You can change this password at any time under the Configuration tab by clicking on Users and editing your existing settings.

    Creating Users and Lines with XiVO

    For those migrating from the FreePBX® world, you’re probably most familiar with the procedure for creating extensions. More advanced administrators may have switched to device and user mode where users and devices are created separately. Phone numbers or extensions were associated with users while phone instruments were associated with devices. In the World of XiVO, we’ll start with the simplest configuration, and you can move on from there when you’re ready. In our scenario today, we’ll create a couple of users. Each user has a Name, Language, Time Zone, and other optional characteristics such as a Mobile Phone Number which can ring simultaneously whenever a user receives a call to his or her local XiVO phone number. By adding a Line (aka Phone Number) for the user as the user account is created, XiVO will automatically generate a separate Line with username and password credentials. This Line will be associated with the User during the initial user setup procedure, and this Line then can be registered to a SIP phone, softphone, or XiVO client (which we will cover separately down the road). In the example below, we’re using Nerd Uno’s extension 701 (associated with line 3jz8tsr0) to call Nerd Dos’ extension 702 (associated with line 8fmne2x4).

    XiVO has an excellent tutorial that covers creating Users with a SIP Line. So jump there and add a couple of Users following the steps in the tutorial. When you’re finished, you’ll have two Users and two associated Lines with credentials to set up SIP phones. Since you’re just getting your feet wet and will probably make some mistakes, it’s probably a good idea to turn off Fail2Ban while you’re experimenting. Otherwise, you may accidentally lock yourself out of your server (ask us how we know) and think it’s a problem with XiVO. Here’s how:

    /etc/init.d/fail2ban stop

    To set up your SIP phones, you’ll need the credentials for each of the two lines. Under the Lines tab, click on the Pencil icon to reveal the Username and Password. Fill in the missing pieces as shown below and make certain that your NAT entry is set to Yes.

    With those credentials in hand, go ahead and configure a couple of SIP phones and make certain you can call between them with audio in both directions before proceeding. For those with a Mac, Telephone is perfect for experimentation because you can set up multiple softphones and place calls between them.

    IMPORTANT: If your server is sitting behind a NAT-based firewall, you must set the external and local network IP addresses for XiVO in General Settings -> SIP Protocol. You’ll find the fields in the Network tab.

    Configuring a SIP Trunk for Google Voice with XiVO

    Now that you have internal calls working, let’s turn our attention to connecting your PBX to the rest of the world. We obviously can’t cover the setup for every SIP provider, but we can provide a good example that will get our U.S. friends free calling in the U.S. and Canada. We’ve chosen the Simonics SIP Gateway to Google Voice because a one-time payment of $5.99 gets you a traditional SIP trunk to interface with any existing Google Voice number. If you don’t have a Google Voice number, sign up here. In your Google Voice Settings, make sure Forward Calls to Google Chat is enabled and disable Call Screening in the Calls tab. Then, with your Google credentials and Google Voice number in hand, visit the Simonics web site to sign up for service. Sign in with your Google credentials and complete the registration process. Once you have your Simonics account name and password, log into your XiVO web portal.

    With credentials in hand, on the XiVO side, start by choosing the SIP Protocol tab under Trunk Management. There are actually three tabs to configure for the SIP trunk. Begin in the General tab and make it look like this using your credentials. NOTE: The complete FQDN for the Simonics gateway should be gvgw.simonics.com:

    Next, click on the Register tab and reenter your credentials. Leave the empty fields exactly as shown. Be sure the Register box is checked.

    Next, in the Signaling tab, change the Monitoring option to Yes and then click Save. Monitoring is the XiVO equivalent of the SIP Qualify option.

    We also need to make one minor adjustment in the SIP Protocol Defaults in the General Settings. Just Save your settings after checking Match users with ‘username’ field.

    Next, we need to tell XiVO how to process Incoming and Outgoing Calls using the Google Voice SIP trunk. Under the Call Management section, let’s begin with the Incoming Calls setup by creating a new Incoming Calls DID for your 11-digit Google Voice number. To keep things simple, we’ll route the incoming calls to the User mapped to extension 701:

    For Outgoing Calls, we need to route calls with a specific dial string out the Simonics SIP trunk using the to-extern context. By way of example, we’ve set this up using a dialing prefix of 48 (GV) and a 10-digit number. We’re letting XiVO supply the missing 1 country code required by Google Voice, and we’ll let XiVO strip off the 48 prefix in processing the outbound calls. If this is your only outgoing trunk, you may prefer not to use a dial prefix at all. In that case, change the dial string to a 10-digit number (NXXNXXXXXX) and set Stripnum to 0.

    Well, that’s enough for today. There’s complete XiVO PDF Documentation available here. We’ll have lots more to say about XiVO in coming weeks. Come join the party!

    Continue reading Part 2.

    Published: Thursday, May 5, 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…

    1. There’s a glitch in the CloudAtCost builds for Debian8. Regardless of how much disk storage you allocate, CloudAtCost will only use 10GB. Moral: Don’t waste your resources by allocating more than 10GB of disk space. This is an experimental platform, and 10GB will suffice. If you really need more space, this thread on the PIAF Forum will walk you through expanding the storage allocation beyond the 10GB threshold. [↩]

    Smartphone Trifecta: 2016’s Very Best Cellphones with Two Awesome Surprises

    Thursday, March 31, 2016


    Every year we try to check out the latest and greatest smartphones with emphasis on finding those that are the best fit with Asterisk®. So this year is really special because our three favorite new phones all come with a couple of surprises. First, monthly cellular service can be FREE on all of them! Second, all of the phone numbers associated with the three phones can be used as free SIP trunks with Incredible PBX™ or your favorite Asterisk server.

    If you’ve been following Nerd Vittles since early February of this year, then you already know that RingPlus, a Sprint MVNO, is the best bargain on the planet. Over the past six weeks of weekly specials from RingPlus, we’ve managed to update all of our free RingPlus accounts to either unlimited calling, texting, and 2GB of monthly data or 3,000 minutes of calling, 3,000 text messages, and 3GB of data. For anybody (except teenagers) that’s sufficient monthly capacity to do almost anything you’d like to do with a smartphone except stream movies all day.

    We initially showed how to take dirt cheap Boost Mobile and Virgin Mobile prepaid phones and repurpose them for use with RingPlus. Sprint apparently read our article as well because that loophole is going away on April 17. However, you still have time to find one and activate it on RingPlus following our previous tutorial. The only catch is that, if you ever deactivate it, you will lose the ability to reactivate it without first using it with Boost or Virgin for a full year. The landfills will be so happy with all these cellphone bricks because of Sprint’s latest attempt to shoot itself in the foot. We think there also are some legal issues that the FCC needs to address. These phones are sold as "contract-free" when, in fact, there is a very specific and undisclosed contractual requirement. If you don’t keep service with the provider for a year, your phone becomes a brick. In antitrust terminology, it’s called tying. And some would argue that it also constitutes false advertising. We plan to file a complaint and would urge all of our readers to do the same. Here’s a link.

    But enough about the Sprint mentality. It really is legendary, and it’s been the same for 20+ years. We doubt it will ever change unless the entire Sprint management team is replaced. So where do we go from here? Well we decided to upgrade most of our phones to the latest and greatest postpaid phones available, and we wanted to try out our 2016 favorites (pictured above). Here’s some really great news. Samsung’s new Galaxy S7 and S7 Edge as well as Apple’s new iPhone SE work swimmingly with RingPlus as long as you purchase the Sprint-branded models at full retail price from either Best Buy or an Apple Store. Sprint and Target will refuse to sell you one unless you activate it with Sprint in the store. You also can’t buy the Sprint-branded iPhones on line from Apple without activating Sprint service, but that restriction doesn’t apply if you visit an Apple Store.

    It took a week to chase down a Galaxy S7 and almost two weeks to find a Galaxy S7 Edge at a Best Buy store. Don’t believe the store inventory on their web site. Neither of the phones we purchased was shown as available at the locations where we bought them. So you’ll need to call or visit a store at least while the new Galaxy phones remain a scarce commodity. As for the iPhone SE, it went on sale at Apple Stores this morning at 10 a.m. At the Charleston store, I was third in line and both of the people in front of me also were buying the Sprint-branded iPhone SE to use with RingPlus. The Apple sales folks said they had never before seen a run on Sprint phones. Guess why?

    Here’s the drill. Purchase your favorite phone after you read our mini-reviews below. Don’t open the box just yet. Instead, look on the bottom of the box and decipher the IMEI/MEID of your phone. Immediately run that number through the RingPlus Device Checker to be sure it will work on the Sprint network using a RingPlus account. There shouldn’t be a problem with any of these three phones, and all of them come with a Sprint SIM card so you won’t have to worry about obtaining one from RingPlus. Some have reported that the Best Buy phones were locked. We can only surmise that the customer delayed activating the phone with RingPlus which gave Sprint time to block the serial number which Best Buy reported. If this happens to you, we are told that Sprint will unlock the phone once you provide proof that it was purchased at full retail price. If all else fails, Best Buy has a 14-day return policy. Remember, anything is possible when dealing with Sprint.

    Once your phone passes the compatibility check, sign up for a new free RingPlus plan. These plans change weekly and sometimes are only offered for a couple of hours so you may want to hold off on signing up until a deal comes along that meets your requirements. Update: There are a number of excellent promotions at the moment which run through April 5. Our previous article explained in detail how these free plans work. Switching plans typically is limited to those that buy into the annual Member+ program. You can read all about the plans and programs on the RingPlus Community Forum. If you already have a RingPlus account with a registered phone, you can swap out the phone with one of these three new ones for a one-time charge of $1.99. All you’ll need is your new MEID and ICC ID numbers. The entire phone swap only takes a minute or two. Once it’s complete, turn on your phone. The rest is automagic!

    Comparing the Phones. We don’t often glow about reviews, but the TechRadar review of the Galaxy S7 Edge is a must-read. There has never been a better phone than this one. And, only an inch behind it is the Galaxy S7 which bears an uncanny resemblance to the new iPhone SE except for its 50% larger screen size. We actually are more comfortable carrying the Galaxy S7 with its all-metal construction. For whatever reason, the S7 Edge always feels like its about a millisecond away from slipping out of your hand. You will most definitely want a case for the S7 Edge.

    In terms of performance and camera quality, the new Galaxy phones are in a league of their own. Here’s a photo hurriedly snapped through a restaurant window with our Galaxy S7 earlier this week. If you’ve ever tried to take sunset pictures with an iPhone or cheapo Android device, you’ll appreciate what a challenge these shots can be. We’ll annotate this article with an iPhone SE photo if and when the opportunity presents itself. The other good news with the new Galaxy phones is they are at least waterproof for a few minutes. If you live near the water, that will come as a welcome addition as well. Finally, Samsung has closed the gap with Apple’s iPhones on backing up and restoring everything on your phone. For years, this has been Apple’s best feature in our humble opinion. Now Samsung goes Apple one better. If you happen to have two Samsung devices that you want clone, simply choose Backup and Reset from Settings. Then Open Smart Switch on both devices and hold the two phones back to back. It’s that easy. Or you can opt for the more traditional restore method that works precisely as it does with an iPhone using the Samsung Cloud. For some additional tips and tricks, visit the PCMag.com site and watch the video "Exploring the Galaxy S7″ which includes a number of comparisons with Apple iPhone devices including the iPhone SE. Enjoy!

    We previously covered the SIP setup for RingPlus devices using their WiFi Fluidcall feature. It provides a free SIP trunk for Asterisk at a cost of zero dollars. For the complete tutorial, take a look at the original article. Enjoy!

    Originally published: Thursday, March 31, 2016



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

     

    Special Thanks to Our Generous Sponsors


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

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

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

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

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

    Some Recent Nerd Vittles Articles of Interest…

    The Sensational Raspberry Pi 3 Featuring Incredible PBX GUI with Raspbian 8 Jessie

    Saturday, March 26, 2016 / 13 Comments on The Sensational Raspberry Pi 3 Featuring Incredible PBX GUI with Raspbian 8 Jessie


    [iframe-popup id="3″]
    Hard to believe it’s been four years since the introduction of the original Raspberry Pi®. Over eight million RasPi’s have been shipped. To celebrate its fourth birthday, Eben Upton has done it again. Meet the sensational Raspberry Pi 3 sporting a 1.2GHz 64-bit quad-core ARM Cortex-A53 CPU with ten times the performance of the original Raspberry Pi. Of particular interest to the VoIP community will be the RasPi 3’s integrated 802.11n wireless LAN and Bluetooth 4.1 hardware. And, of course, the RasPi 3 retains its compatibility with the Raspberry Pi 1 and 2. Did we mention it’s still just $35? Because we like to celebrate birthdays, too, we’re pleased to introduce a brand new Incredible PBX™ image for the Raspberry Pi 2 and 3 featuring Raspbian 8 and the latest release of Asterisk® 13. Unlike previous builds, this one installs in under a minute. Yes, it’s still FREE and features pure open source GPL code. No Gotchas!

    07/01/2019 NEWS FLASH: Just released Incredible PBX LITE for the Raspberry Pi 2, 3, and 4 featuring Raspbian 10 Buster. Tutorial here.

    Raspberry Pi 3 Performance. Gone are the days of worrying about Raspberry Pi performance. Both the user interface and call quality now match what you’d expect to find on a $300-$500 VoIP server. Even with a Raspberry Pi 2, we have detected no performance degradation thanks to the latest Raspbian 8 OS and a virtually flawless Asterisk 13 platform. For best results, we recommend 32GB Class 10 microSD cards which now are plentiful at the $10 price point.1

    Raspberry Pi 3 Shopping List. Before you can install Incredible PBX, you’ll need a compatible Raspberry Pi 3 platform. Here’s the short list:

  • $35* Raspberry Pi 3 from MCM or Newark or Amazon
  • $10 Power Adapter (2.5 amps minimum!)
  • $9 32GB microSDHC Class 10 card
  • £12.95 Rainbow Pibow case or $9.50 Official RasPi case

    • About That Asterisk. We write about Asterisk® regularly, but the asterisk we’re talking about is the one accompanying the $35* price tag for the Raspberry Pi 3. Yes, that’s the advertised price. And, no, if you want one this year, you’re not going to pay that. There are the marked up shipping prices, the bundled add-on’s that you don’t need or want, and the must-have accessories like a power adapter. We’re assuming you already own a USB keyboard and an HDMI-compatible monitor. If so, just plan on $100 and consider yourself lucky if you get all the pieces for less. Our order from Pimoroni in the U.K. with a case and 3-day shipping was £59.36 or $82.95 U.S. Our order from MCM for just the RasPi 3 with shipping was $46.99.

    Incredible PBX Feature Set. Where to begin? Let’s start with the Alphabet Stew: IAX, SIP, GVSIP, SMS, and SRTP functionality. Voice Recognition and Text-to-Speech VoIP application support using FLITE, GoogleTTS, and PicoTTS. Free calling with Google Voice, Simonics SIP gateway, or RingPlus cellular service. And all of your Nerd Vittles favorites: Fax, AsteriDex, Click-to-Dial, News, Weather, Reminders, and Wakeup Calls. Plus hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Email Delivery of Voicemail, Voicemail Blasting, and more…

    10-Layer Network Security Model. Most phone calls cost money. Unlike many of the other "free" VoIP solutions, our most important criteria for VoIP is rock-solid security. If your free server ends up costing you thousands of dollars in phone bills due to fraud, it isn’t free at all. Once you plug in that network cable, you’ve painted a bullseye on your checkbook.

    No single network security system can protect you against zero-day vulnerabilities that no one has ever seen. Deploying multiple layers of security is not only smart, it’s essential with today’s Internet topology. It works much like the Bundle of Sticks from Aesop’s Fables. The more sticks there are in your bundle, the more difficult it is to break them apart. If a vulnerability suddenly appears in the Linux kernel, or in Asterisk, or in Apache, or in your favorite web GUI, you can continue to sleep well knowing that other layers of security have your back. No one else in the telecommunications industry has anything close. Ours is all open source GPL code so we would encourage everyone to get on board and do their part to make the Internet a safer place!

    Do your part and do your homework. Comparison shop as if your phone bill matters! 😉 Incredible PBX provides:

    1. Preconfigured IPtables Linux Firewall
    2. Preconfigured Travelin’ Man 3 WhiteLists
    3. Randomized Port Knocker for Remote Access
    4. TM4 WhiteListing by Telephone (optional)
    5. Fail2Ban Log Monitoring for SSH, Apache, Asterisk
    6. Randomized Ultra-Secure Passwords
    7. Automatic Update Utility for Security & Bug Fixes
    8. Asterisk Manager Lockdown to localhost
    9. Apache htaccess Security for Vulnerable Web Apps
    10. Security Alerts via RSS Feeds in Kennonsoft and Incredible PBX GUIs

    Installation Tutorial. Here’s everything need to know about installation and setup. "Automatic" means you just watch.

    1. Download and unzip Incredible PBX image from SourceForge (with or without GV OAuth support)
    2. Transfer Incredible PBX image to microSD card
    3. Boot Raspberry Pi from new microSD card
    4. Login to RasPi console as pi:raspberry to initialize your server (Automatic)
    5. Reboot after writing down your server IP address (Automatic)
    6. Login via SSH as root:password to randomize passwords & configure firewall (Automatic)
    7. Optionally, install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)
    8. Enjoy!

    Configuring Trunks with Incredible PBX

    Before you can actually make and receive calls, you’ll need to add one or more VoIP trunks with providers, create extensions for your phones, and add inbound and outbound routes that link your extensions to your trunks. Here’s how a PBX works. Phones connect to extensions. Extensions connect to outbound routes that direct calls to specific trunks, a.k.a. commercial providers that complete your outbound calls to any phone in the world. Coming the other way, incoming calls are directed to your phone number, otherwise known as a DID. DIDs are assigned by providers and you register your trunks using credentials handed out by these providers. Incoming calls are routed to your DIDs which use inbound routes telling the PBX how to direct the calls internally. A call could go to an extension to ring a phone, or it could go to a group of extensions known as a ring group to ring a group of phones. It could also go to a conference that joins multiple people into a single call. Finally, it could be routed to an IVR or AutoAttendant providing a list of options from which callers could choose by pressing various keys on their phone.

    We’ve done most of the prep work for you with Incredible PBX. We’ve set up an Extension to which you can connect a SIP phone or softphone. We’ve set up an Inbound Route that, by default, sends all incoming calls to a Demo IVR. And we’ve built a dozen trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re good to go.

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

    With the preconfigured trunks in Incredible PBX, all you need are your credentials for each provider and the domain name of their server. Log into Incredible PBX GUI Administration as admin using a browser. From the System Status menu, click Connectivity -> Trunks. Click on each provider you have chosen and fill in your credentials including the host entry. Be sure to uncheck the Disable Trunk checkbox! Fill in the appropriate information for the Register String. Save your settings by clicking Submit Changes. Then click the red Apply Config button.

    Configuring a Softphone for Incredible PBX

    We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP telephony.

    We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You’ll need the IP address of your server plus your extension 701 password. Choose Applications _> Extensions -> 701 and write down your SIP/IAX Password. You can also find it in /root/passwords.FAQ. Fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password you assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

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

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

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

    Configuring Google Voice

    If you want to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

    IMPORTANT: Do NOT under any circumstances take Google’s bait to switch from Google Chat to Hangouts, or you will forever lose the ability to use Google Chat with Incredible PBX. Also be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. Good News! You’re in luck. Google has apparently had a change of heart on discontinuing Google Chat support so it’s enabled by default in all new Google Voice accounts. Once you’ve created a Gmail and Google Voice account, go to Google Voice Settings and click on the Calls tab. Make sure your settings match these:

    • Call Screening – OFF
    • Call Presentation – OFF
    • Caller ID (In) – Display Caller’s Number
    • Caller ID (Out) – Don’t Change Anything
    • Do Not Disturb – OFF
    • Call Options (Enable Recording) – OFF
    • Global Spam Filtering – ON

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

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

    Now you’re ready to configure your Google Voice account in Incredible PBX. You can do it from within the Incredible PBX GUI by choosing Connectivity -> Google Voice. How you enter your credentials depends upon whether you have chosen the Incredible PBX image with OAuth 2 support. For a complete Google Voice OAuth tutorial, follow steps 8-10 in this Nerd Vittles tutorial. Once you’ve entered your credentials, you MUST restart Asterisk from the command line, or Google Voice calls will fail.

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

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

    Another option is to use an inexpensive SIP Gateway to Google Voice. The Simonics trunk in the Incredible PBX GUI is preconfigured for this purpose. All you’ll need is your Google Voice credentials. Get started with this tutorial.

    Adding Speech Recognition Support to Incredible PBX

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

    To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Do NOT change the yourID portion of the command: 

    sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747

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

    To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. "What planes are overhead." Read the Nerd Vittles tutorial for additional examples and tips.

    Enabling WiFi on the Raspberry Pi

    With the Raspberry Pi 3, wi-fi hardware is included. With the Raspberry Pi 2, you’ll need to add an inexpensive wifi dongle. The next step is connecting to your WiFi router. Simply open /etc/wpa_supplicant/wpa_supplicant.conf with your favorite editor and insert the following code using the actual SSID name and password to access your local, password-protected WiFi router or any open WiFi network:

    network={
 ssid="YourSSID"
 psk="YourSSIDpassword"
 key_mgmt=WPA-PSK
 scan_ssid=1
 priority=5
}

network={
 key_mgmt=NONE
 priority=1
}

    Finally, stop and restart the wlan0 interface, count to 15, and check the status of your server to decipher the new IP address for your WiFi connection:

    ifdown wlan0
ifup wlan0
pbxstatus

    If you want to run your Raspberry Pi exclusively off the WiFi connection, simply unplug the network cable from your RasPi and reboot your server.

    UPDATE: There still is a quirk with the wireless LAN driver on the Raspberry Pi 3. The problem has to do with the default power management of the wlan0 interface which results in it being powered off after very brief periods of inactivity. Special thanks to Matt Gemmell for this fix. Just cut-and-paste the lines below into a terminal window, and you’ll be good to go.

    WARNING: Run pbxstatus first. If the top line shows Raspberry Pi 3, the following WiFi patch is already installed.

    echo "options 8192cu rtw_power_mgnt=0 rtw_enusbss=0 rtw_ips_mode=1" > /etc/modprobe.d/8192cu.conf
sed -i '/exit 0/d' /etc/rc.local
echo "sleep 10" >> /etc/rc.local
echo "iwconfig wlan0 power off" >> /etc/rc.local
echo "exit 0" >> /etc/rc.local
echo "[Unit]" > /etc/systemd/system/root-resume.service
echo "Description=Turn off wlan power management" >> /etc/systemd/system/root-resume.service
echo "After=suspend.target" >> /etc/systemd/system/root-resume.service
echo "" >> /etc/systemd/system/root-resume.service
echo "[Service]" >> /etc/systemd/system/root-resume.service
echo "Type=simple" >> /etc/systemd/system/root-resume.service
echo "ExecStartPre= /bin/sleep 10" >> /etc/systemd/system/root-resume.service
echo "ExecStart= /sbin/iwconfig wlan0 power off" >> /etc/systemd/system/root-resume.service
echo "" >> /etc/systemd/system/root-resume.service
echo "[Install]" >> /etc/systemd/system/root-resume.service
echo "WantedBy=suspend.target" >> /etc/systemd/system/root-resume.service
systemctl enable root-resume
reboot

    After rebooting, if you issue the iwconfig wlan0 command, it should show: Power Management:off.

    Update: Lessons Learned for Raspberry Pi 3 Road Warriors

    As with all new devices, you learn some things as you go along. So we’re providing an update to our original article to offer a couple of additional tips and tricks for those that want to travel with a RasPi…

    Alternative Power Sources. If you’re like us, you have a number of devices around the house or office that all require 5V power adapters of various amperages. The Raspberry Pi has traditionally been one of the most temperamental when it came to power adapters and, with the Raspberry Pi 3, the developers specifically mention a 2.5 amp minimum. If you travel and want to take devices such as the Raspberry Pi with you, the last thing you want to do is approach airport security with a bunch of wires hanging out of your carry-on bag. Well, there’s good news. The Anker device shown in the Amazon ad in the right column of Nerd Vittles can supply power to 6 devices including a Raspberry Pi 3. And we’ve given the RasPi a healthy workout with no adverse effects.

    Deciphering the RasPi IP Address. As we mentioned, we travel a lot so obtaining a DHCP address for your RasPi in WiFi mode is not always the easiest thing to accomplish. If your smartphone supports tethering, that’s the easiest way to get connectivity on the road. A better way is to stick a WiFi HotSpot in your luggage and it, too, can be powered using the Anker device. See our recent article for WiFi HotSpot choices. Regardless of which option you choose, it will require some planning to use your RasPi sans monitor and keyboard. First, you need to preconfigure /etc/wpa_supplicant/wpa_supplicant.conf with the SSID of the device you’ll be using to hand out DHCP addresses. You’ll note from the discussion above that each entry in this file has a priority with higher numbers having higher priority. The way we typically do this is to assign our home network as the highest priority. Below that, we set up credentials for our MiFi Hotspot, then our smartphones, and finally open networks. So it looks like this:

    • Home Network – 6
    • MiFi HotSpot – 5
    • Android phone – 4
    • iPhone (AT&T) – 3
    • Open Network – 1

    Keep in mind that the Incredible PBX firewall probably will block you from accessing the RasPi from a computer on the public network. So you also must connect your computer to the same private WiFi network because private LAN addresses are whitelisted in the firewall by default.

    Once you have connectivity for your RasPi and your laptop, the other wrinkle is figuring out the IP address of the Raspberry Pi. Our recommended approach goes like this. First, configure SendMail on the RasPi to use a Gmail account that you own as an SMTP smarthost to send emails. That should work almost anywhere you go. Second, modify /etc/rc.local to automatically send you an email with the IP address and SSID of your wireless network whenever the RasPi boots. Again, this takes some advance planning because you need to set all of this up and test it before you go on the road.

    Here are the steps to modify SendMail to use an existing Gmail account as a SmartHost. Log into your RasPi as root and issue the following commands:

    cd /etc/mail
hostname -f > genericsdomain
touch genericstable
makemap -r hash genericstable.db < genericstable
mv sendmail.mc sendmail.mc.original
wget http://nerdvittles.dreamhosters.com/pbxinaflash/source/sendmail/sendmail.mc.gmail
cp sendmail.mc.gmail sendmail.mc
mkdir -p auth
chmod 700 auth
cd auth
echo AuthInfo:smtp.gmail.com \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" > client-info
echo AuthInfo:smtp.gmail.com:587 \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" >> client-info
echo AuthInfo:smtp.gmail.com:465 \"U:smmsp\" \"I:user_id\" \"P:password\" \"M:PLAIN\" >> client-info
nano -w client-info

    When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.

    Now issue the following commands. In the last step, press ENTER to accept all of the default prompts:

    chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
sed -i 's|sendmail-cf|sendmail\/cf' /etc/mail/sendmail.mc
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/Makefile
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/databases
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc.gmail
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf.errors
sendmailconfig

    Next, edit /etc/hosts and /etc/hostname. Change the raspberypi3 entries to: raspberrypi3.incrediblepbx.com.

    Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!

    /etc/init.d/sendmail stop
/etc/init.d/sendmail start
apt-get install mailutils -y
echo "test" | mail -s testmessage yourname@yourdomain.com

    The last step is to add these commands to /etc/rc.local to send you an email with your IP address and SSID whenever the RasPi is rebooted. Insert the following commands just above the exit 0 line at the end of the file. Use an email address to which you have access on the road!

    ESSID=`iwconfig | grep ESSID | tail -1 | cut -f 9 -d " "`
echo "IP address: $(hostname -I) on $ESSID" | mail -s "RaspberryPi3 IP Address" yourname@yourdomain.com

    Enabling Bluetooth on the Raspberry Pi


    Incredible Fax Returns for the Raspberry Pi


    Mastering the Incredible PBX Feature Set

    Now would be a good time to explore the Incredible PBX applications. Continue reading there. If you have questions, join the PBX in a Flash Forums and take advantage of our awesome collection of gurus. There's an expert available on virtually any topic, and the price is right. As with Incredible PBX, it's absolutely free. Enjoy!

    Originally published: Monday, March 7, 2016  Updated: Saturday, March 26, 2016

    Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It's the best Asterisk tech support site in the business, and it's all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won't have to wait long for an answer to your question.


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

     

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

    I Have A Dream: Free Cellular Service with Integrated Remote SIP Connectivity

    Monday, February 8, 2016 / 6 Comments on I Have A Dream: Free Cellular Service with Integrated Remote SIP Connectivity

    As part of our Mobile Internet adventure this year, we’ve been scouring the countryside with two requirements in mind. First, we wanted a smartphone on which we could activate some type of free cellular service for making calls and sending text messages. Second, we wanted to integrate remote SIP connectivity using the same provider and phone number so that we could make and receive calls transparently using any SIP phone or Asterisk® server anywhere in the world. Sounds like a tall order, you say? Well, if you’ve enjoyed your Cloud@Cost Sandbox, you’re gonna love RingPlus!

    Yes, you’ll have to buy a compatible cellphone, but there are thousands to choose from. And, yes, you’ll need Sprint service in your neighborhood. Then you’ll have to cough up $10 to activate your cellular account. RingPlus offers dozens of plans.1 We recommend the Michelangelo plan which best meets what we’re trying to accomplish today, but the choice is all yours.2 With the Michelangelo plan, you can make and receive 1,000 minutes of free calls a month to anywhere in the U.S. (calls to Canada are 3¢ a minute), you can send and receive 1,000 free text messages a month, and you can use 500MB of free data service every month. You also can use your same account credentials with any SIP phone, softphone, or Asterisk server anywhere in the world to make and receive phone calls transparently using the same phone number as your smartphone. In other words, you can travel anywhere and make and receive phone calls just as if you were sitting in Atlanta, Georgia dialing from your smartphone. The SIP calls are deducted from your free minutes. No cellular service required at all. Meet RingPlus!


    So what’s the catch? How does RingPlus make money? Well, of course, they would prefer that you sign up for a plan with monthly fees. For those on the free plans, the only difference you will notice is an occasional ad which plays instead of a ring tone when you place outbound calls. This only occurs until the other party answers the call, and it can be all but eliminated by choosing a music selection in the RingPlus Radio feature in your RingPlus Dashboard.

    Who are the ones most likely to use something like this? Well, for openers, all of your kids unless you like springing for a $500 phone and spending $40+ dollars a month for cellular service for each of them. One of the other real beauties of RingPlus is you can set up a whitelist of numbers that can be called from the phone. Blacklists are supported as well. It’s perfect for kids just getting started with a cellphone. A second potential user group would be those who travel outside the United States and prefer not to pay exorbitant roaming rates for calls. Using a SIP phone connected to your RingPlus account, all of the international calls suddenly are free. And the calls are delivered with the same CallerID number as calls placed from your actual smartphone. In fact, your smartphone doesn’t have to be in service at all. A third and perhaps most important use for us was to serve as a failover trunk on one or more Asterisk servers. When all else fails, you can route outbound calls to your RingPlus SIP trunk for free calling using your RingPlus account. Doesn’t get any better than that.

    Official RingPlus WARNING: Starting April 17, 2016, per our carrier partner Sprint, Members and potential Members will no longer be able to activate prepaid devices which are not eligible under Sprint’s FED policies [Requires activation of prepaid phone on original Sprint MVNO network for at least one year!]. Such prepaid devices will no longer pass FED until actual eligibility date is met.

    There are probably numerous ways to put all these pieces in place so that things function just as we’ve described. Today we’ll share with you the solution that actually worked for us. You can take it from there and avoid the thousands of horror stories about incompatible smartphones. Be advised that acquiring used cellphones or even incompatible cellphones is a very dangerous and expensive business. If you buy one that happens to be stolen, or that has a balance due on the account, or that is incompatible with RingPlus, then you’ve bought a tiny boat anchor and not much else. So, our best advice is buy one from the provider. That’s the one and only RingPlus, and the smartphones start at just under $100. Many Sprint post-paid phones also work, such as the new iPhone SE (Sprint Model) from any Apple Store.

    If store employees will let you, find the Sprint postpaid phone that you like and look on the bottom of the box. There you will find the decimal value of the MEID. Log into http://nerd.bz/nvringplus and plug in the MEID to see if it is RingPlus compatible. If it passes, buy it. If it flunks, try another one. Whatever you do, DON’T BUY A PHONE IN AN OPENED BOX, AND DON’T OPEN THE BOX YET! Make certain there is a return policy in case things don’t work out as expected!

    Funny story. The Radio Shack employees at our local store were very savvy and refused to let me look at the MEID claiming it was a security issue. Fair enough. Of course, they were also curious why I wanted a phone without letting them configure it. Once I told them the deal, they all wanted one, too. They asked for the link to the MEID verification site and said they’d do it for me. Once it worked, excitement broke out in the room with all the staff reading an early copy of this article. While Radio Shack typically charges a $35 restocking fee on cell phones, that fee is waived if you return the phone in an unopened box. So the only thing you’re wasting if they insist that you purchase the phone is a little bit of your time and a lot of Radio Shack employee time if, in fact, the MEID flunks the verification test.

    Configuring Your Phone for RingPlus Service

    Now sign up for a RingPlus free plan using the MEID and ICC ID you previously verified. Michelangelo is probably the best bet if you missed our Twitter tip this past weekend. Deposit $10 in your new account, and activate it. Log into your RingPlus Dashboard, click on your phone in the upper right frame, and choose Manage Device. Write down your MSID, your phone number, and MSL. Once your account is active, then and only then unbox and turn on your phone. Go through the minimal setup steps by choosing your Language and choosing an available WiFi network. During this setup, RingPlus should push a PRL update to your new phone, and it will reboot. Check in Settings -> General -> About Phone -> Status and see if you have a phone number. If so, you’re good to go. If not, open the Phone Dialer application and dial ##72786# which should force another PRL update to your phone with another reboot. When it finishes, check again for a phone number and place an outbound call.

    Using a browser on your desktop computer, go back into the RingPlus Dashboard and sign in. Your phone device should show Active in the upper right corner of the screen. Click there and you’ll get a display like this:

    While still in the Device Settings Menu, click on the WiFi FluidCall option to decipher your SIP credentials. You’ll need these to set up your SIP phone or a SIP trunk on your Asterisk server. Your username is your 10-digit phone number, the domain name is sip.ringplus.net, and the password is a system-generated entry which you can recreate whenever you like. That’s probably a very good idea whenever you use public WiFi services to make calls with your SIP phone or a softphone.

    By the way, this isn’t some kludgy SIP-GSM gateway where the calls actually are routed out through your cellphone device. The RingPlus SIP gateway connects your SIP device directly to the Internet and simply uses your existing RingPlus CallerID to identify the calls. In short, you get the best of both worlds: a dirt cheap or free cellphone service plus a dirt cheap or free SIP trunk for use anywhere in the world.

    Configuring a RingPlus SIP Trunk with Asterisk

    If you’d like to set up your RingPlus number as a failover trunk on your Asterisk server, here is the setup that worked for us with Incredible PBX using your assigned 10-digit phone number for your username and fromuser settings and your assigned password for your secret. If you include a registration string and configure an inbound route using your RingPlus DID, then inbound calling will work as well. If you skip the registration step, then you can use the same RingPlus trunk on multiple Asterisk servers for emergency outbound calling. No firewall adjustments should be necessary.

    There are all sorts of other magic tricks you can implement using the RingPlus API, but you probably won’t need any of the features in light of the robust SIP connectivity RingPlus provides to an existing Asterisk server where the feature set is virtually unlimited. Be advised that you must make a call out at least once every 60 days to keep your account active. The simple way to do this is to set up a monthly reminder using your RingPlus trunk. Schedule the reminder to call out once every month using Telephone Reminders in Incredible PBX.

    RingPlus Gotcha Checklist

    Free service wouldn’t be free without a few land mines. So here’s a checklist to keep things running smoothly without any problems down the road. First, link your account to one of the social media options (Twitter, Facebook, or LinkedIn) when you sign up for service. You’ll find the link on your Dashboard under the Your Social Networks icon. Second, make at least one outbound call a month on every line you activate. As noted, this can be accomplished automatically using the Telephone Reminders application in Incredible PBX. Third, keep a valid credit card on file in your account at all times. Fourth, keep a positive balance in your account for each phone that you activate to avoid automatic replenishment at the original rate when you signed up for your plan. Fifth, be mindful of the Domino Effect. With some plans, if you allow a related plan to end (for example, Queen of Hearts when you also have an Ace of Hearts plan), then your better plan will be demoted in its feature set. Enjoy the Free Ride!

    Originally published: Monday, February 8, 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…

    1. Be advised that future upgrades of these "free" plans may go away after February 15 unless you join the Member+ program, the cost of which changes almost weekly. This will not affect those that already are participating in the program according to RingPlus. [↩]
    2. In case you’re curious, a plan equivalent to the free Michelangelo plan at RingPlus would run you $41.00 per month at Ting. Ouch! [↩]

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