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Clearly Anywhere: The Ultimate Mobile User VoIP Companion

The wait is over! If you like sliced bread, then today’s your lucky day to enjoy the next best thing. Perhaps the single most important component in the VoIP toolbox for any PBX deployment that includes mobile users is a softphone for Android and iOS devices. We began this month with a tip for users of a limited number of Android smartphones. And we have previously written about Linphone and Zoiper and Telephone. We’ve also sung the praises of earlier releases of Google’s Pixel smartphone which makes a perfect VoIP companion even without a cellphone provider. Today we’re pleased to introduce Clearly Anywhere for Incredible PBX 2020 PUBLIC platforms that checks all the boxes: security, reliability, flexibility, feature set, emergencies, and a very reasonable price that covers unlimited devices per user extension. Downloading Clearly Anywhere onto your smartphone from the App Store or Google Play, setup is instantaneous using Clearly Anywhere’s native QR code activation.

  • Security: SIP credentials need not be shared with the Softphone provider
  • Reliability: Incoming calls arrive via Push Notifications whether app is running or not
  • Flexibility: With PJsip, multiple iOS & Android devices can share the same extension
  • Feature Set: SMS, BLFs, recording, video calls, DND per phone with Custom Scheduling
  • Emergencies: Easily reroute 911 calls to the cell carrier for reliable location identification
  • Intro Price: Through October 31: $9.99 per user per year with grandfathered rate lock




 

Clearly Anywhere Setup on Incredible PBX 2020

Before you deploy Clearly Anywhere on your smartphones, you first must set things up on the PBX side. Here’s how the process works. First, you will need to deploy the PUBLIC version of Incredible PBX 2020. This will reconfigure the IPtables firewall to support Clearly Anywhere. It requires assignment of a fully-qualified domain name (FQDN) for your server. The FreePBX® Clearly Anywhere module is licensed on a per PBX basis. In order to activate the FreePBX module, you’ll next need to purchase one or more user licenses from our Incredible PBX Store. As part of the purchase process, you’ll be prompted to provide the FQDN of your Incredible PBX 2020 PUBLIC server plus the TCP port that Clearly Anywhere clients will use to communicate with your PBX. By default, that port is TCP 2267. We will whitelist TCP port 2267 (or a port of your choice) as part of this setup process. Finally, you will be provided a Registration Token to activate the Clearly Anywhere module when you first run it in FreePBX. Then you simply activate the Clearly Anywhere client for the extensions corresponding to the number of licenses you have purchased.

1. Let’s begin by installing the Clearly Anywhere module for FreePBX. Log into your server as root and issue the following commands from the CLI. NOTE: After this initial install, future updates to the Clearly Anywhere module will be handled in the traditional way using the Module Admin utility in FreePBX.

fwconsole ma downloadinstall http://incrediblepbx.com/clearlysp-14.0.2.tgz
fwconsole reload
/root/sig-fix
/root/sig-fix

2. Purchase one or more client licenses from the Incredible PBX Store. Keep in mind that, if multiple users will be sharing access the same extension, you only need a single license for each extension. And, if you will be setting up multiple smartphones using the same extension on your PBX, that only requires a single license as well.

3. When prompted, provide the FQDN of your server as well as the TCP port that will be used by Clearly Anywhere clients to communicate with your PBX. Default is TCP 2267. To avoid service interruptions, we strongly recommend against using a dynamic IP address for your server. Even with a dynamic DNS utility, there will be times when your Clearly Anywhere clients cannot reach the PBX unless it has a static, public IP address. If you need a cloud-based solution, the $25/year offering from Crown Cloud has no equal.

4. Decipher the Registration Token from the Clearly IP email confirmation.

5. Using a web browser, access the FreePBX GUI using your admin credentials. Navigate to Settings -> Clearly Anywhere. Enter your Registration Token at the prompt. Your Clearly Anywhere configuration will display. NOTE: If you purchase additional client licenses down the road, revisit this page and click Refresh to update your license count.

6. Next, you need to activate Clearly Anywhere for each PJsip extension that will be used with the softphone clients. Each additional extension requires a separate Clearly Anywhere license. Navigate to Admin -> User Management and then Edit the desired Username. Click on the Clearly Anywhere tab and set Enable Softphone to ENABLE. If you previously have subscribed to Incredible SIP Trunking in our ClearlyIP Store, then you also have the option of enabling SMS Messaging on any of the DIDs assigned to your account. As the video above explains, this SMS messaging carries across all of the Clearly Anywhere clients as well as the UCP connected to this same extension on your PBX. IMPORTANT: In PJsip Advanced settings, remember to increase the Max Contacts entry by two for each softphone that will be connecting to the PJsip extension.

7. Whitelist TCP 2267 assigned for Clearly Anywhere API client connections to your server:

cd /etc/sysconfig
sed -i 's/10000:20000 -j ACCEPT/&\\n-A INPUT -p tcp -m tcp --dport 2267 -j ACCEPT/' iptables
iptables-restart

Clearly Anywhere Setup on Smartphones

The setup process on the smartphone side is simple. Begin by downloading the Clearly Anywhere app from either the App Store or Google Play for your device. Once the client softphone is installed, run the app. You will be prompted for the UCP user credentials and FQDN of your PBX. You can manually enter these, or you can access the User Control Panel (UCP) for the desired user/extension and activate the QR code Provisioning widget for the account. Then, instead of entering the credentials manually on every softphone, you can choose the QR code option and simply point your smartphone at the QR code displayed in UCP. Connections are virtually instantaneous.NOTE: The beauty of this design is SIP credentials need not be shared with the provider.1 While UCP accounts, by default, are created using the extension numbers to which they are associated, these usernames can be changed to anything you like. So, for the 701 extension, if you want to change the UCP login name in your extension settings to BettyBoop, that’s perfectly okay.

Once connected, click on the Settings icon in Clearly Anywhere. Verify that Push Notifications are enabled for Incoming Calls. In the Preferences tab, set a Ring Tone for incoming calls and decide whether to record calls. The Network preference by default will choose WiFi and then Cellular with automatic roll-over when needed. If you want 911 calls routed through your cell carrier instead of through your PBX, choose Number Rewriting, click +. For Conditions, choose Equals 911. For Actions, choose Override Dial Action and select gsmCall. Click Done.

You’re now a Clearly Anywhere Expert. We hope you enjoy using the Ultimate Softphone with Incredible PBX 2020.
 

Originally published: Monday, September 21, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



  1. If you activate Push Notifications, SIP credentials must be stored on the SIPIS server in order to reregister your softphone with your PBX when incoming calls arrive. This is an Asterisk and PJsip limitation. []

Frozen in Time: Sangoma at a Crossroads or the Cliff

We are fast approaching the 18 month anniversary since the mass exodus of talent from Sangoma and Digium. During that time, Asterisk® and FreePBX® have languished in limbo with FreePBX commercial modules mired in discontinued encryption methods and the Asterisk and FreePBX feature set frozen. Sangoma’s acquisition of Digium gave the company a well-respected business phone system in Switchvox® while relegating FreePBX deployments with commercial modules to second-class citizen status.

Now Sangoma has a difficult and expensive decision to make. In order to continue to deploy FreePBX commercial modules on a current operating system and PHP platform, they are facing a multi-million dollar investment to retool all of the commercial modules using an entirely different encryption platform which will generate additional headaches with their module repositories which now must manage and maintain two different sets of modules. Think of it as migrating from an English-based operation to one in which the customer base speaks either English or French, but not both. In the meantime, Asterisk 17 is fast approaching end-of-life while Asterisk 16 is getting long in the tooth with only minor fixes of late. And there’s FreePBX 15 which hasn’t seen significant upgrades since the departure of the former Schmoozecom staff in April of 2019.

While we have never agreed with the Sangoma CEO about much of anything, let’s take over the management reins for a moment and look at the landscape. The Digium acquisition coupled with some other purchases has left Sangoma with a mountain of debt in the middle of a global pandemic. Coronavirus also has forced many businesses to reassess their long-term office strategy. With nearly everyone on the planet now carrying a smartphone and most current employees having high-speed Internet connectivity at home neither of which the company has to pay for, does a fancy office for every employee still make good business sense? When you have a well-positioned business phone offering in Switchvox, does it make good business sense to continue to pump boatloads of money into a competing FreePBX commercial product simply to preserve its functionality? At the same time, Sangoma’s open source Asterisk and FreePBX products need a serious shot in the arm to remain viable. For example, FreePBX 15 with its User Control Panel won’t run on CentOS 8, and CentOS 7 reaches EOL status in December.

So here’s our first act as Sangoma CEO. Convert all of the existing FreePBX commercial modules into open source, GPL code while offering existing FreePBX commercial module customers a 50% discount to migrate to Switchvox within the next six months. We would stagger conversion of the commercial modules to GPL code by releasing three on the first day of every month. This would keep Sangoma in the news for many months to come. The move would not only restore Asterisk and FreePBX to their well-deserved position at the forefront of the open source community, but it would also give Switchvox a leg up in the commercial phone system marketplace. By keeping the identity of the monthly releases a secret, it would build much needed momentum for Sangoma’s open source offerings while reinforcing Sangoma’s commitment to the open source community. Give it some thought, Sangoma!
 

Originally published: Tuesday, September 8, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



Android Alert: Unmasking Your Hidden SIP Phone

Perhaps the single most important component in the VoIP toolbox for any PBX deployment that includes traveling users is the softphone whether it’s deployed on notebook computers or smartphones. We’re kicking off football season with reviews of some of our favorite SIP softphones for mobile users. We’ve previously written about Linphone and Zoiper and Telephone. We’ve also sung the praises of earlier releases of Google’s Pixel smartphone which makes a perfect VoIP companion even without a cellphone provider. Today we’re passing along an Android tip from @w1ve on the VoIP-Info.org Forum for anyone needing PBX connectivity while away from the home or office. And it won’t cost you a dime so long as you already have an Android smartphone with either a Wi-Fi connection or a data plan with any cellphone carrier.

You may be unaware that Android has been shipping with a native SIP phone since as far back as Android Marshmallow which will be five years old next month. Some phone manufacturers such as Chinese-owned Lenovo/Moto disable the SIP functionality, but many do not including Google, Samsung, and OnePlus. If you’ve deployed an Incredible PBX 2020 PUBLIC server in the Cloud with a PJsip extension, the beauty of this discovery is that you’ll always have VoIP connectivity through your PBX with only a WiFi connection. No SIM is required!

To determine if your Android smartphone includes the SIP dialer, simply open the native Phone app and tap the three vertical bar icon at the top of the dialer menu. From the pull-down menu, choose Settings, Calls, and Calling Accounts. If your Android smartphone includes support for the native SIP dialer, there will be a SIP Accounts option in the menu.

TIP: Even if the SIP Menu doesn’t appear on your smartphone, it may be that the manufacturer simply disabled the menu without actually trashing the SIP functionality as Lenovo has done. To determine whether the menu has simply been hidden, install QuickShortcutMaker from the Play Store. Run the app and scroll down and tap the SIP Settings option and click Test. It should bring up the SIP Accounts menu shown below.

Tapping SIP Accounts and then + icon will open a dialog to add a SIP account to your phone.


With an Incredible PBX 2020 PUBLIC server, the entries should look like the following:

  • Username: Your PJsip extension
  • Password: Your PJsip password
  • Server: FQDN of Your PBX
  • Optional Settings -> Port: 5061

Keep in mind that an FQDN for the Server address is required with an Incredible PBX 2020 PUBLIC server. If your server is using a dynamic IP address, you also would want to configure the FQDN using a Dynamic DNS service and refresh that FQDN periodically on your PBX using a cron job. Once you’ve entered your credentials, tap SAVE to activate the SIP account on your smartphone. It should then appear in the SIP Accounts window as shown above.

Next, you have some choices to make as to how the SIP account is actually used. As you can see from our setup (shown above), we allow outbound calls using either the SIM card or the SIP phone, and the phone will prompt for a choice whenever you make a call. We also have activated Inbound SIP calling which, as the dialog explains, uses some battery life. Finally, if you elect to use a chan_SIP extension on your PBX, make certain that you have enabled NAT Mode in the Advanced tab, or you will experience one-way or no audio on calls. This is not required with PJsip extensions. One of the other beauties of PJsip extensions is that you can assign this extension to multiple SIP devices including softphones and desktop phones so long as you increase the Max Contacts entry in the Advanced tab for the PJsip extension. In this way, you can answer incoming calls on your desktop phone when you’re at home or in the office and answer the same calls on your smartphone when you’re out and about. Enjoy!
 

Originally published: Tuesday, September 1, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



A New World: Adding Cellular Extensions to Incredible PBX


Over the past few weeks, we’ve introduced a revolutionary new technology for the Asterisk® and FreePBX® community by marrying Incredible PBX® with GTI Global cellular extensions. For $20 to $35 a month, you get a new SIM card for your unlocked GSM cellphone that links all four U.S. carrier networks to a PJsip extension on your PBX. Incoming calls ring on both the cellphone and any other SIP phone connected to the same PJsip extension. Outbound calls from the cellphone get processed just like every other outbound call on your PBX. All the other bells and whistles that make Asterisk and FreePBX a one-of-a-kind platform work exactly the same way on the smartphone as they do with every other PBX extension. And your cell phone is no longer tied to a single carrier’s network. Based upon your current U.S. location, the GTI SIM finds the strongest signal by automatically selecting from the cell towers of the four major U.S. carriers: AT&T, Verizon, T-Mobile, and Sprint. And last week, we added additional icing to the cake by introducing the perfect cloud platform for Incredible PBX in either Atlanta or Los Angeles for a jaw-dropping $25 a year with a 5-minute install.

Today we want to document the setup procedure to put all the cellular pieces in place to add a GTI-enabled cellphone to your Incredible PBX platform. In a nutshell, there are preliminary configuration steps on your PBX so that GTI can verify that your PBX can be successfully integrated into their cellular network. Next, you’ll need to sign up for service with GTI Global and provide some basic information about your server configuration. Once you receive credentials from GTI Global, you’ll need to complete some additional setup steps on your PBX including creation of a GTI trunk as well as configuring inbound and outbound routes to support your cellphone calls. You’ll also need to add exceptions for each GTI DID to the [from-sip-external] context. Finally, await delivery of your SIM card and then decide how you will deploy the GTI SIM on a GSM cellphone of your choice.

GTI Global provides promotional consideration to help defray costs of our Asterisk projects.

Preliminary Configuration Steps with Incredible PBX

To connect a GTI-enabled cellphone to Incredible PBX, here are the preliminary steps:

  1. Incredible PBX must have a static IP address on the public Internet.
  2. Whitelist a number of GTI IP addresses in your firewall.
  3. Create a PJsip extension for each GTI-enabled cellphone.
  4. Create a Preliminary Chan_SIP Trunk for Registration to GTI.

These whitelist entries should be added to iptables-custom in the /usr/local/sbin directory:

/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.122.166 --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.110.51  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 96.127.174.38  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.170  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.171  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.172  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.173  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s someid.mvnoserver.com --dport 5060:5069 -j ACCEPT

 
For each GTI cellphone you plan to deploy, add a PJsip extension with a very secure password:

Extension: 484xx1
Display Name: 484xx1 GTI
Secret: yourSuperSecretpassword

Advanced -> Max Contacts: 3

 
Also make note of the SIP port that is configured for your PJsip extensions, typically 5061. It’s shown as Port to Listen On under Settings -> SIP Settings -> PJsip Settings in FreePBX GUI.

Create a preliminary Chan_SIP Trunk for registration to GTI leaving the CallerID field empty:

Trunk Name: trunk4321

PEER Details:

username=trunk4321
type=friend
secret=yourBIGpassword
host=xxxxxx.mvnoserver.com
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=5062

Register string: trunk4321:yourBIGpassword@xxxxxx.mvnoserver.com:5062

 

Registration Information to be Provided to GTI

As part of your GTI setup procedure, in addition to your name and a reachable cellphone number, you will be prompted for the following information. If any of it is missing, open a support ticket and provide it. NOTE: you must register through our portal, or your request will be denied unless you pay the standard $250 setup fee which is waived for our users. All GTI accounts through Nerd Vittles include one free SIM card, a free trunk account to process calls to and from your GTI pjSIP extension(s), and a free DID for each SIM card. Unlimited calling and unlimited SMS/MMS messaging is included on all plans. Pricing difference is for monthly data allocations: $20 (no data), $25 (2GB), and $35 (4GB). You must pay for the first month’s service BEFORE your SIM card will be shipped.

  1. PBX Public IP Address and Trunk Listening Port (5062)
  2. Extension IP address (PBX public IP address)
  3. pjSIP Extension Number
  4. pjSIP Extension Secret
  5. pjSIP Extension Port (5061)
  6. Service Desired: $20, $25, or $35 monthly plan
  7. Area Code Desired for DID
  8. Business Name and Shipping Address for Your SIM

Configuring Your PBX with GTI Credentials

Once GTI has verified that they can communicate with your PBX, you will be provided the credentials for your GTI Trunk. Simply edit the trunk you configured above inserting the Trunkname/Username, Secret, Host, and Port for your new Trunk. Modify the Register String, save your settings, and reload your dialplan. Also edit your whitelist and change the FQDN from someid.mvnoserver.com to the host name that was provided. Then restart the firewall: iptables-restart.

Verifying Successful Extension & Trunk Registrations. Once your GTI extension and trunk are configured. You can verify successful connections in the Asterisk CLI.
For the extension: pjsip show aors
For the trunk: sip show registry

Configuring an Inbound Route for GTI Global Calls. Replacing 16785551212 with your actual 11-digit DID assigned to each cellphone, create Inbound Routes that look like this:

Description: GTIglobal 16785551212
DID Number: 16785551212
Destination: Extension -> 484001 (extension number of matching SIM)

 
Also add the following context at the end of /etc/asterisk/extensions_custom.conf using your DID. Then reload your dialplan: asterisk -rx "dialplan reload"

[from-sip-external]
; GTI Global
exten => 16785551212,3,Goto(from-trunk,${DID},1)

 

Configuring an Outbound Route for GTI Global Calls. No special outbound routes are required unless you wish to use the GTI Global trunk to process U.S. calls at no cost. In this case, configure an Outbound Route pointing to your GTI Global Trunk. NOTE: Calls using this trunk can only be made from the extensions associated with GTI Global SIMs.

CallerID and Trunking Strategy. In setting up the GTI components in FreePBX, you need to consider how you wish to process outbound calls. If you want all outbound calls including those from the GTI extensions to reflect the company’s main phone number, then you can use your default outbound route making certain that CallerID override is set with the company number in the appropriate trunks. No CallerID should be set in the outbound route.

If you want calls from the GTI extensions to use the GTI trunk with the CallerID number of the individual cellphones, then the GTI trunk needs to be moved to the top of the default route sequence with no CallerID number specified in the GTI trunk or the outbound route. Instead, set the correct 11-digit CallerID numbers in each of the GTI extensions. This will not impact outbound calls from other extensions because the GTI trunk will indicate congestion forcing the calls out through the next trunk specified in the outbound route.

I’ve Got My GTI SIM. Now What?

It’s decision time on what sort of smartphone you wish to use with your GTI SIM. If you’re a heavy data consumer then it may make sense to acquire a cellphone that supports dual SIMs and use a SIM with an unlimited data plan as your data provider and use the GTI SIM for business purposes with your PBX.

Choosing a Cellphone Platform. Our favorite deployment strategy is to take advantage of the new dual SIM offerings that allow a user to have one SIM for personal use and a second SIM (GTI) for business use. This lets you acquire the least expensive cell service with unlimited data for personal use and acquire the $20 GTI SIM with no data plan through our Incredible PBX offering. You don’t really need two data plans on the same smartphone. A couple of options worth exploring include GTI’s MVNO offering of T-Mobile unlimited data service for $35 a month and Visible (a Verizon-owned MVNO) that offers unlimited plans for as little as $25 a month with a four-user signup. The users need not be related, and the users can each pay their own monthly bill. Keep in mind that MVNOs almost always have fine print allowing "deprioritization" in busy areas and during busy times of the day. And the underlying carriers always screw their MVNO customers first.

Some smartphones support two SIMs while others have one physical SIM slot plus an eSIM registration that can be acquired from the major carriers. We personally liked an unlocked $200 Motorola G8+ which has two actual SIM slots as do most of the newer Samsung, OnePlus, Huawei, and many other smartphones. Several cellphones from BLU are available for under $25. The following devices currently support a single SIM card plus an eSIM:

  • Apple® iPhone® SE
  • Apple iPhone 11
  • Apple iPhone 11 Pro
  • Apple iPhone 11 Pro Max
  • Apple iPhone XR
  • Apple iPhone XS
  • Apple iPhone XS Max
  • Google Pixel 4 / Pixel 4 XL

Configuring the API Settings for GTI. Once you insert your SIM into your smartphone and power it on, you need to modify the API settings and choose GTI Global (Mobile-X) as your provider for the GTI SIM. Here is a FAQ covering the appropriate API settings depending on whether you’re using an iPhone or Android phone. Here’s another tip. Even though your phone may show a number of available carriers other than Mobile-X, that doesn’t mean you can connect to any of them directly. Why? Because you don’t have credentials to access those networks. Your connection to those other networks is only available through Mobile-X using your GTI credentials.

Testing Out Your GTI Smartphone

Once you complete all of the steps above, it’s time to try things out. From your cellphone, try dialing *98 to connect to Asterisk voicemail. Next, try calling another extension on your PBX. Then try calling an outside number. Finally, try calling the DID of your GTI SIM from another phone and be sure it rings and you have audio in both directions.

When you complete the testing, you can add another SIP phone to the same extension as your GTI cellphone. Be sure to set the port to the proper port for your PJsip extension. When you call your GTI SIM’s DID, both the smartphone and the SIP phone should ring simultaneously.

In some environments, you may wish to configure 611 calls to ring a support ring group in your organization. If so, add a new SIP extension 611 and forward calls to that extension to the number of the desired ring group. For example, if the Ring Group number is 777, you could enter the following command at the Asterisk CLI: database put CF 611 777.
 

Originally published: Monday, July 13, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



Last Chance to Jump onto Incredible PBX Cellular Bandwagon




We introduced our new GTI Technologies cellular extensions for U.S.-based Incredible PBX® platforms last week, and the enthusiastic response has been truly amazing. This week we want to share the results of our testing, and encourage you to become one of the pioneers for this exciting new technology. It can be integrated into almost any Asterisk® platform.

To briefly summarize, GTI cellular extensions provide a traditional cellphone connection to all four of the major U.S. carriers. GTI-enabled cellphones also register as traditional PJsip extensions with Asterisk. The GTI SIM deciphers the best network based upon the cellphone’s current location and then processes inbound and outbound calls just as if your cellphone were a registered desktop SIP phone on your PBX. Incoming calls are routed according to the inbound routing rules on your PBX, and outgoing calls follow the outbound routing rules of your PBX. 911 calls are processed directly by the carrier to whom your cellphone is connected.

GTI Global provides promotional consideration to help defray costs of our Asterisk projects.



This is not our first rodeo, and we wander into new communications territory with our eyes wide open. So the real test of this new platform was our recent 1,000 mile voyage in the southeastern United States. Our benchmark cellphone is registered on the AT&T network and reported one or two bars of coverage throughout much of the journey. That’s more than adequate to handle calls, text messaging, Waze, and TuneIn Radio with less than a handful of momentary outages. But what a difference four carrier connectivity makes. On our Google Pixel 3 with the GTI SIM, we observed four-bar cellular coverage throughout most of the trip. So my first observation would be that those in remote locations with spotty cellphone service are likely to experience a dramatic improvement in connectivity thanks to access to almost every cellphone tower in the United States. This has nothing to do with Asterisk and everything to do with simply having a better cellphone experience in general.

Now for the Asterisk component using a GTI SIM-based cellphone. If you write down the top ten things you love about Asterisk, you now can do almost all of them from your cellphone: transfer calls, listen to your voicemail, call other extensions and ring groups, use various trunking options by dialing prefixes to save on international calls, chatting with colleagues, unlocking a front door, opening a gate, retrieving news and weather data, and on and on. You can even use the traditional 611 cellular option to route calls to your help desk or security services. And, thanks to the magic of PJsip, folks can call your office extension and both your cellphone and desktop phone ring. They’ll never know you’ve left the office.

Pricing for GTI cellular service is competitive until you get into the high data categories. Unlimited calling and texting is available on all of the plans. With no data service which is not required for calling and messaging, the monthly cost is just $20. A 2GB data plan is $25.

And here is an update on an Unlimited Data Plan that we received this morning:

We are working on building an "unlimited" data plan that is based on a data throttling mechanism. This mechanism would look like this:

User would be sold a 20GB data bundle. Once the user reaches 15GB, the data channel speed would be throttled to 128 kbps. This should stretch the remaining 5GB of data so it will last through the end of the month. While it is not truly unlimited data [nobody else has it either], it gives the perception of unlimited data to the end-user. This type of approach is under review by the business team.

UPDATE: GTI Global is a T-Mobile MVNO. So another option would be to purchase a separate T-Mobile SIM to cover data requirements. An unlimited data T-Mobile SIM costs $35/month while an unlimited plan with 2GB 4G LTE data and 3G throttling runs $20/month. There is a $2 charge for the additional T-Mobile SIM.

And here is another update. Canada DIDs are now available for an additional $0.55 per month. Cellphone calls still have to be initiated or received within the U.S.

Choosing a Cellphone Platform. Our favorite deployment strategy is to take advantage of the new dual SIM offerings that allow a user to have one SIM for personal use and a second SIM (GTI) for business use. This lets you acquire the least expensive cell service with unlimited data for personal use and acquire the $20 GTI SIM with no data plan through our Incredible PBX offering. You don’t really need two data plans on the same smartphone.

Some smartphones support two SIMs while others have one physical SIM slot plus an eSIM registration that can be acquired from the major carriers. We personally liked an unlocked $200 Motorola G8+ which has two actual SIM slots as do most of the newer Samsung, OnePlus, Huawei, and many other smartphones. Several cellphones from BLU are available for under $25. The following devices currently support a single SIM card plus an eSIM:

  • Apple® iPhone® SE
  • Apple iPhone 11
  • Apple iPhone 11 Pro
  • Apple iPhone 11 Pro Max
  • Apple iPhone XR
  • Apple iPhone XS
  • Apple iPhone XS Max
  • Google Pixel 4 / Pixel 4 XL

Become a Pioneer. At the end of this week, we’ll submit to GTI the email addresses of those wanting to venture into the future along with us. We think the companies you represent and the customers you support will thank you six months from now. GTI service is rock-solid, the technology has no competition, and the rewards for you and the organizations you support are noteworthy. You must have an Incredible PBX server or comparable Asterisk server with a static, public Internet address to use this service. Setup details are in last week’s article.

If you’d like to join the adventure, please shoot us an email request this week and keep us informed about your progress. We can’t wait to share in your discoveries!
 

UPDATE: After you hear back from us with the signup details, here’s what happens next. You will sign up on our GTI Portal. Within an hour or two, you will get an email from GTI requesting the IP address of your server and extension credentials for each GTI SIM. Once you have provided that information, GTI will provision your account. It typically takes about 2 to 4 hours for service setup on carriers network (and some time less that 30 minutes). Then you will receive further notice/request and or status of the progress via a support ticket with a copy to your email address. GTIGLOBAL IP address and FQDN information will be provided after GTI has completed the setup of your Asterisk/FreePBX domain and IP in GTI’s system. If GTI needs clarification or additional information, GTI will contact you via support ticket. Once provisioned, all of the information will be available in your GTI portal. We’re providing the sequence of events so that you can be on the lookout for each step. If it doesn’t happen, open a support ticket.

Continue Reading: Soup-to-Nuts Tutorial: Adding Cellular Extensions to Incredible PBX
 

Originally published: Monday, June 29, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



Introducing Cellular Extensions for Incredible PBX & Asterisk



The Holy Grail for many of us in the VoIP community has been a cellphone that functions like a traditional Asterisk® extension on your PBX. The typical use case would be a real estate agent, service technician, salesperson, or other mobile worker who interacts with a home office as part of their daily routine. The beauty of this for the mobile workforce is it allows both the home office receptionist and the mobile worker to not only exchange calls but also to transfer the calls and retrieve voicemail just as if the worker were using a phone in the office.

Some of you may recall that we introduced a service like this called vMobile from Vitelity about six years ago. vMobile had some growing pains not the least of which was total reliance upon Sprint for cellular coverage. The service has been discontinued.


Today we’re testing a new offering from Global Technologies (GTI Global) on the Incredible PBX® platform that addresses some of the shortcomings of the vMobile offering. First, it provides a SIM card that can automatically utilize all four of the major carrier networks in the United States. Second, it offers a dedicated DID, a SIP trunk with free U.S. calling, and an Asterisk extension for every cellphone. And third, it takes advantage of PJsip which permits multiple phones to be connected to and ring on the same Asterisk extension.

GTI Global provided promotional consideration to help defray costs of our Asterisk projects.

Prerequisites. System integrators will acquire SIM cards through the Incredible PBX project. Pricing for the recommended unlimited calling, unlimited messaging, and 2GB monthly data plan is $25/month. A NoData plan is available for $20/month, and other higher data plans also are available. The SIM is compatible with any iPhone® or Android phone device. Users must provide their own cellphones which need not be jailbroken. Dual-SIM smartphones allow users to dedicate one line to the GTI Global SIM and another line for personal use. GTI Global currently is exploring support for eSIMs which would facilitate iPhone use.

To use the SIM-based phone, a public-facing Asterisk platform with a dedicated IP address is required, and it must support PJsip extensions. Incredible PBX 2020, Incredible PBX 16-15, and Incredible PBX 13-13 all have been tested. Other FreePBX® 13, 14, and 15 platforms should also suffice. Accompanying each SIM is a dedicated DID assigned to each cellphone. A matching PJsip extension must be created on the Asterisk platform, and the SIP credentials must be provided to GTI Global to make the cellphone connection to the PBX. Additional phones may be connected to the same PJsip extension to support a desktop phone or receptionist. Simply adjust the Max Connects entry when creating the PJsip extension to support the number of phones desired. Each phone on an extension requires a unique port.

How Calling Works. GTI Global has positioned servers in the facilities of every carrier. When a call arrives from a carrier’s tower, the carrier processes 911 calls directly through its network. For all other incoming calls, the carrier verifies the credentials of the SIM. Once verified, the call is passed to the GTI Global server which sends incoming calls to the associated extension on your PBX. The Inbound Route associated with that DID then sends the call to the assigned destination. If the GTI Global server is unavailable, the carrier processes the incoming calls just as it would any other call on its network by sending it directly to the cellphone.

When the cellphone user places a call, it is processed just as if the call had been made from an internal extension on the PBX with the exception of 911 calls which are handled directly by the carrier. After SIM verification, GTI Global passes the outgoing call to the PBX for processing using the Outbound Routing rules of the PBX. Internal extensions and voicemail can be dialed directly. Outbound calls can utilize any trunk associated with the PBX including the free U.S. trunk provided by GTI Global. CallerID is determined by the outbound trunk processing the call.

Configuring a GTI Global Extension. After logging into the FreePBX GUI, create a PJsip extension with the following settings:

Extension: 12345
Display Name: 12345 GTIglobal
Secret: yourSIPpassword

Advanced -> Max Contacts: 4

Configuring a GTI Global Trunk. Create a chan_SIP Trunk in the FreePBX GUI using your GTIglobal-provided credentials:

Trunk Name: trunk4321

PEER Details:

username=trunk4321
type=friend
secret=yourTRUNKpassword
host=xxxxxx.mvnoserver.com
disallow=all
allow=ulaw
canreinvite=no
dtmfmode=rfc2833
port=5062

Register string: trunk4321:yourTRUNKpassword@xxxxxx.mvnoserver.com:5062

IMPORTANT: When you sign up with GTI, you need to give them the IP address of your PBX, the credentials (extension and password) for your GTI PJsip extension, the outbound SIP port for your GTI trunk (UDP 5062), and the inbound PJsip port for your PJsip extension (UDP 5061).

Configuring an Inbound Route for GTI Global Calls. Replacing 16785551212 with your actual 11-digit DID assigned to each cellphone, create Inbound Routes that look like this:

Description: GTIglobal 16785551212
DID Number: 16785551212
Destination: Extension -> 12345 (your extension associated with each SIM)

You need to add the following context at the end of /etc/asterisk/extensions_custom.conf using your DID. Then reload your dialplan: asterisk -rx "dialplan reload"

[from-sip-external]
; GTI Global
exten => 16785551212,3,Goto(from-trunk,${DID},1)

Configuring an Outbound Route for GTI Global Calls. No special outbound routes are required unless you wish to use the GTI Global trunk to process U.S. calls at no cost. In this case, configure an Outbound Route pointing to your GTI Global Trunk. NOTE: Calls using this trunk can only be made from the extensions associated with GTI Global SIMs.

CallerID and Trunking Strategy. In setting up the GTI components in FreePBX, you need to consider how you wish to process outbound calls. If you want all outbound calls including those from the GTI extensions to reflect the company’s main phone number, then you can use your default outbound route making certain that CallerID override is set with the company number in the appropriate trunks. No CallerID should be set in the outbound route.

If you want calls from the GTI extensions to use the GTI trunk with the CallerID number of the individual cellphones, then the GTI trunk needs to be moved to the top of the default route sequence with no CallerID number specified in the GTI trunk or the outbound route. Instead, set the correct 11-digit CallerID numbers in each of the GTI extensions. This will not impact outbound calls from other extensions because the GTI trunk will indicate congestion forcing the calls out through the next trunk specified in the outbound route.

GTI Global Firewall WhiteList. On Incredible PBX servers and other IPtables platforms, add the following whitelist entries to your firewall:

/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.122.166 --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 99.198.110.51  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 96.127.174.38  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.170  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.171  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.172  --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp -s 37.18.129.173  --dport 5060:5069 -j ACCEPT

How to Proceed. If GTI Global cellular extensions are of interest for your customers, kindly contact us at support@incrediblepbx.com, and we’ll hook you up with the folks at GTI Global. There also is an extensive KnowledgeBase for those wanting more information.

Continue Reading: Last Chance to Jump onto Incredible PBX Cellular Bandwagon
Soup-to-Nuts Tutorial: Adding Cellular Extensions to Incredible PBX
 

Originally published: Monday, June 22, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



Free Cellular Service in the U.S. & Canada Returns

We’re a few months late to the party but, if Covid-19 has put a kink in your cellular budget or if you have young kids that could use a cellphone with unlimited calling and texting on the Sprint® network at zero cost in the U.S. and Canada, then your prayers have been answered. All you need is a CDMA phone and a $10 SIM card from TextNow. Better yet, you can purchase one of their refurbished iPhones or Android phones, and you’re off to the races. Their compatible web app with 200+ million downloads lets you make calls and send text messages from your favorite desktop computer or tablet. Special thanks to @Hometech for this tip.

What’s the Catch? The free tier is ad-supported so you’ll get a few ads. But, other than that, unlimited calling and texting throughout the U.S. and Canada are free whether you’re on WiFi or on Sprint’s nationwide network. If the ads are bothersome, you can always move up to the $10/month ad-free platform which also gets you voicemail transcription. There’s even free data service albeit on Sprint’s 2G network. But it’s perfectly suited for email. Yes, we’re aware of Sprint’s terrible coverage in some areas, but hopefully this will improve with the T-Mobile merger. Much like Google Fi, the TextNow design supports WiFi wherever it’s available. You can’t beat the price, and no credit card is required to sign up. So what are you waiting for?

Referral Code Saves $20. TextNow no longer offers referral codes or affiliate links, and we don’t have one. But here’s a link to some existing coupons that will save you $20 if you decide to purchase a phone from them.


 
Free Features: In addition to a free phone number and unlimited calling and texting to U.S. and Canadian numbers, TextNow offers conference calling, voicemail, and group texting. Calling and texting via the web app or platform-specific applications is just as intuitive and powerful as using a cellphone. In fact, you can set things up on your existing phone while you’re waiting on your TextNOW SIM card or smartphone to arrive. Come join the party!
 

Originally published: Friday, May 22, 2020



Need help with Asterisk? Visit the VoIP-info 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.
 



Introducing Incredible PBX LITE featuring Raspbian 10



As you may know, the Raspberry Pi Foundation introduced the $35 Raspberry Pi 4 last week. In addition to jaw-dropping hardware enhancements, the introduction also included the new Raspbian 10 (Buster) platform which was surprising since Debian 10 isn’t scheduled for official release until the end of this week. As with prior releases, Raspbian 10 brought with it some major headaches for the VoIP community not the least of which is FreePBX® cannot (yet) be installed. But sometimes there’s a silver lining accompanying bad news.

Several Debian 10 issues caused us to rethink what a VoIP platform for the Raspberry Pi actually should look like. After all, most Raspberry Pi hobbyists aren’t interested in mastering the intricacies of Asterisk® and FreePBX. They’re more concerned with a stable, fast VoIP communications platform that’s easy to deploy and will operate without hiccups in a home or small office environment. Thus was born Incredible PBX LITE, a new turnkey VoIP platform that requires zero configuration out of the box and supports five SIP telephones and up to four trunk providers for low-cost worldwide calling. Simply sign up with one of these VoIP service providers, acquire a telephone number (DID), enter the IP address of your PBX, and you can instantly make and receive calls using up to five SIP telephones or softphones.

UPDATE: If you’d prefer the full-featured Incredible PBX 16-15 for the Raspberry Pi, it’s now available here as well.

What’s Included? Despite its name, Incredible PBX LITE still serves up a VoIP powerhouse featuring Asterisk 13, an Apache web server, the latest MariaDB SQL server (formerly MySQL), SendMail, and most of the Incredible PBX feature set including SIP, SMS, Opus, voice recognition, FLITE Text-to-Speech VoIP applications plus fax support, Click-to-Dial, News, Weather, Reminders, and hundreds of features that typically are found in commercial PBXs: Conferencing, IVRs and AutoAttendants, Email Delivery of Voicemail, Voicemail Blasting…

What’s Missing? We’ve removed the entire FreePBX GUI platform while retaining most of its feature set. We’ve also eliminated the need to run a web server or database server although they’re still there. And gone are the days of having to configure extensions and trunks as well as inbound and outbound routes before you can actually use your PBX to make your first call. The tradeoff is a noticeable performance improvement. While a Raspberry Pi 4 isn’t required to run Incredible PBX LITE, doing so provides another three-fold performance boost compared to a Raspberry Pi 3B+. Simply stated, Incredible PBX LITE performance now rivals what you would expect on a powerful cloud-based platform such as Digital Ocean or Vultr.

Choosing a SIP Provider. As we mentioned, Incredible PBX LITE comes preconfigured to support four of the major SIP providers: Skyetel, VoIP.ms, V1VoIP, and Anveo Direct. We obviously hope you’ll choose Skyetel not only because they financially support Nerd Vittles and our open source projects, but also because it is a clearly superior platform offering crystal-clear communications and triple-redundancy so you never miss a call. Skyetel also sets itself apart from the other providers in the support department. They actually respond to issues, and there’s never a charge. As the old saying goes, they may not be the cheapest, but you get what you pay for. Even without taking advantage of Nerd Vittles half-price offer on up to $500 of Skyetel services, they’re still dirt cheap compared to the Bell Sisters and cable companies. Traditional DIDs are $1 per month. Outbound conversational calls are $0.012 per minute. Incoming conversational calls are a penny a minute, and CallerID lookups are $0.004. With all four providers, you only pay for minutes you use. Using more than one is a good idea. Effective 10/1/2023, $25/month minimum spend required.


Assembling the Required Raspberry Pi Components

Before you can deploy Incredible PBX LITE, you’ll first need the necessary Raspberry Pi hardware. Here’s the short list and, if you’re in a hurry, the $35 Raspberry Pi 3B+ will cost you less than $3 extra to get it quickly from Amazon using our referral link. If you prefer to wait for a Raspberry Pi 4, read on. Either way, the RasPi remains one of the world’s best bargains! Assuming you already own an HDMI-compatible monitor and a USB keyboard

  • Raspberry Pi 4B from a Raspberry Pi reseller
  • $8 15.3W USB-C RasPi 4 (only) Power Supply
  • $8 32GB microSDHC Class 10 card (strongly recommended!)
  • $5 Official RasPi 4 Case
  • Getting Started with Incredible PBX LITE

    Here’s everything to know about installation and setup. "Automatic" means just watch. Steps #1 and #2 are self-explanatory. For the remaining steps, we’ll further document the procedures in the sections below.

    1. Download and unzip Incredible PBX LITE image from SourceForge
    2. Transfer Incredible PBX LITE image to microSD card
    3. Boot Raspberry Pi from new microSD card (16GB minimum)
    4. Login to RasPi console as root:password to initialize your server (Automatic)
    5. In raspi-config Advanced Options, Expand FileSystem to fill your SD card
    6. In Localization Options, set Locale, TimeZone, Keyboard, & WiFi Country
    7. Reboot after writing down your server IP address (Automatic)
    8. Login via SSH or Putty as root:password to set passwords & setup firewall (Automatic)
    9. Register for and configure at least one trunk provider for Incredible PBX LITE
    10. Install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)

    First Boot of Incredible PBX LITE with Wi-Fi

    Incredible PBX LITE requires Internet connectivity to complete its automated install. If you’re using a wired network connection, you can skip to the next section. With the Raspberry Pi 3B and 4B, WiFi is built into the hardware. But you still have to insert your SSID name and SSID password to make a connection to your WiFi network. To do so, follow these next steps carefully. Insert the Incredible PBX LITE microSD card into your Raspberry Pi 3 or 4 and apply power to the hardware. When the bootup procedure finishes, login as root with the default password: password. At the first prompt, DO NOT PRESS THE ENTER KEY! Instead, press Ctrl-C to break out of the setup script. At the command prompt, issue the following commands to bring up the WiFi config file:

    cd /etc/wpa_supplicant
    nano -w wpa_supplicant.conf
    

    If your WiFi network does not require a password, then uncomment the four lines below and save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    network={
     key_mgmt=NONE
     priority=1
    }
    

    If your WiFi network requires a password, scroll down to the SSID entry and replace YourSSID with the actual SSID of your WiFi network. Make sure you preserve the entry with the quotes as shown. Next, replace YourSSIDpassword with the SSID password of your WiFi network. Save the file: Ctrl-X, Y, then Enter. Now restart your server: reboot. When the reboot finishes, you now should have network connectivity.

    Once the reboot process finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in a minute. Skip the next section since you are using a WiFi connection.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (invalid WiFi credentials, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS.

    First Boot of Incredible PBX Using Wired Connection

    Incredible PBX LITE requires Internet connectivity to complete its automated install. After connecting your server to your local network with a network cable, insert the Incredible PBX LITE microSD card into your Raspberry Pi and apply power to the hardware. When the bootup procedure finishes, you should see an entry on about the middle line displayed on your monitor which reads: "My IP address is…". Write down the IP address shown. You’ll need it in the next step.

    If you don’t see an IP address assigned to your server, then correct the network deficiency (cable not connected, DHCP not working, Internet down), and reboot until you see an IP address assigned to your server. DO NOT PROCEED WITHOUT AN ASSIGNED IP ADDRESS.

    Completing the Incredible PBX Initialization Procedure

    Unless your desktop PC and RasPi are both on the same private LAN, the remainder of the install procedure should be completed from a desktop PC using SSH or Putty. This will assure that your desktop PC is also whitelisted in the Incredible PBX firewall. Using the console to complete the install is NOT recommended as your desktop PC will not be whitelisted in the firewall. This may result in your not being able to log in to your server. Once you have network connectivity, log in to your server as root from a desktop PC using the default password: password. Accept the license agreement by pressing ENTER. You then will be redirected to raspi-config. This is the utility used to expand your Incredible PBX LITE image to use your entire microSD card. If you fail to complete this step, your microSD card will be restricted to 16GB. In the raspi-config utility, choose Localization Options and set Locale, TimeZone, Keyboard, & WiFi Country. Then choose Advanced Options. All of the defaults should be satisfactory with the exception of the first item: Expand Filesystem. Choose this option and activate the resizing directive. Review the other items and then exit and reboot.

    Once your server reboots and you log back in as root, all of your passwords will be randomly assigned with the exception of the root user Linux password. You can set it by issuing the command: passwd. With the exception of your root user password, the remaining passwords can be displayed using the command: /root/show-passwords.

    Finally, if your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060 and UDP 10000-20000 traffic to the private IP address of your PBX. This is required for all of the SIP providers included in the Incredible PBX LITE build. Otherwise, all inbound calls will fail.

    SECURITY ALERT: There was a configuration error in the initial setup which leaves the firewall deactivated. This gets corrected by the Incredible PBX Automatic Update Utility the next time you login to your server as root. Please do so immediately.

    Configuring Skyetel for Incredible PBX LITE

    If you’ve decided to go with Skyetel, here’s the drill. Sign up for Skyetel service and take advantage of the Nerd Vittles BOGO special. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the BOGO credit for your account by referencing the Nerd Vittles special offer. Skyetel will match your deposit of up to $250 which gets you up to $500 of helf-price calling. Credit is limited to one per person/company/address/location.

    Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX LITE:

    • Name: MyPBX
    • Priority: 1
    • IP Address: PBX-Public-IP-Address
    • Port: 5060
    • Protocol: UDP
    • Description: lite1.incrediblepbx.com

    To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.

    Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here.

    Configuring VoIP.ms for Incredible PBX LITE

    To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX LITE server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls.

    Configuring V1VoIP for Incredible PBX LITE

    To sign up for V1VoIP service, sign up on their web site. Then login to your account and order a DID under the DIDs tab. Once the DID has been assigned, choose View DIDs and click on the Forwarding button beside your DID. For Option #1, choose Forward to IP Address/PBX. For the Fowarding Address, enter the public IP address of your server. For the T/O (timeout) value, set it to 2o seconds. Then click the Update button. Under the Termination tab, create a new Endpoint with the public IP address of your server so that you can place outbound calls through V1VoIP.

    Configuring Anveo Direct for Incredible PBX LITE

    To sign up for Anveo Direct service, sign up on their web site and then login. After adding funds to your account, purchase a DID under Inbound Service -> Order DID. Next, choose Configure Destination SIP Trunk. Give the Trunk a name. For the Primary SIP URI, enter $[E164]$@server-IP-address. For Call Options, select your new DID from the list. You also must whitelist your public IP address under Outbound Service -> Configure. Create a new Call Termination Trunk and name it to match your server. For Dialing Prefix, choose six alphanumeric characters beginning with a zero. In Authorized IP Addresses, enter the public IP address of your server. Set an appropriate rate cap. We like $0.01 per minute to be safe. Set a concurrent calls limit. We like 2. For the Call Routing Method, choose Least Cost unless you’re feeling extravagant. For Routes/Carriers, choose Standard Routes. Write down your Dialing Prefix and then click the Save button.

    Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Login to your server as root and use nano to edit extensions_additional.conf in the /etc/asterisk directory. Search (Ctl-W) for anveo-pin and replace anveo-pin with the 6-digit alphanumeric PIN for your account. Press Ctrl-X, Y, then Enter to save your settings. Reload your dialplan with the command: asterisk -rx "dialplan reload"

    Audio Issues with Incredible PBX LITE

    Only if you experience one-way or no audio on some calls, add your external IP address and LAN subnet in /etc/asterisk/sip_general_custom.conf like the following example:

    nat=yes
    externip=xxx.xxx.xxx.xxx
    localnet=192.168.0.0/255.255.0.0 
    

    Then restart Asterisk: systemctl restart asterisk

    Configuring a Softphone for Incredible PBX LITE

    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 can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was 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
    TODAY - Today in History
    LENNY - The Telemarketer's Worst Nightmare
    

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

    Incredible PBX LITE Administration

    We’ve eased the pain of administering your new PBX with a collection of scripts which you will find in the /root folder after logging in with SSH or Putty. Here’s a quick summary of what each of the scripts does.

    add-fqdn is used to whitelist a fully-qualified domain name in the firewall. Because Incredible PBX LITE blocks all traffic from IP addresses that are not whitelisted, this is what you use to authorize an external user for your PBX. The advantage of an FQDN is that you can use a dynamic DNS service to automatically update the IP address associated with an FQDN so that you never lose connectivity.

    add-ip is used to whitelist a public IP address in the firewall. See the add-fqdn explanation as to why this matters.

    del-acct is used to remove an IP address or FQDN from the firewall’s whitelist.

    proximity is a script used in conjunction with bluetooth to decipher whether your smartphone is within range of your server. If not, the script forwards calls to extension 701 to an extension or external smartphone of your choice. Edit the proximity script to add your preferences. Then uncomment the proximity line in /etc/crontab. Complete setup details on setup are available in our previous tutorial.

    reset-conference-pins is a script that automatically and randomly resets the user and admin pins for access to the preconfigured conferencing application. Dial C-O-N-F from any registered SIP phone to connect to the conference.

    reset-extension-passwords is a script that automatically and randomly resets ALL of the SIP passwords for extensions 701-705. Be careful using this one, or you may disable existing registered phones and cause Fail2Ban to blacklist the IP addresses of those users. HINT: You can place a call to the Ring Group associated with all five extensions by dialing 777.

    reset-reminders-pin is a script that automatically and randomly resets the pin required to access the Telephone Reminders application by dialing 123. It’s important to protect this application because a nefarious user could set up a reminder to call a number anywhere in the world assuming your SIP provider’s account was configured to allow such calls.

    show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX LITE actually is.

    show-passwords is a script that displays ALL of the passwords associated with Incredible PBX LITE. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.

    update-IncrediblePBX is the Automatic Update Utility which checks for server updates from incrediblepbx.com every time you log into your server as root using SSH or Putty. Do NOT disable it as it is used to load important fixes and security updates when necessary. We recommend logging into your server at least once a week.

    pbxstatus (shown above) displays status of all major components of Incredible PBX LITE.

    Call Detail Records available in spreadsheet format at /var/log/asterisk/cdr-csv/Master.csv.

    Originally published: Monday, July 1, 2019



    Need help with Asterisk? Visit the VoIP-info Forum.


     

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