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The Most Versatile VoIP Provider: FREE PORTING

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



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



Choosing the Best VoIP Platform for Your Needs

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

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

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

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

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

VoIP Platform Feature Summary

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

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

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

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

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

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

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

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

 

Deploying a Local Server vs. Cloud Platform

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

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


Available Free Trunks for VoIP Servers

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

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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

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

Cloud 9: Free Incredible PBX in the Cloud Hosting until 2019




These deals don’t come along every day so we’re interrupting our regular programming to alert you to a terrific, limited time cloud hosting offer for first-time users of Vultr. If you hurry, you can take advantage of a $25 credit on Vultr which translates into 10 free months of cloud hosting service. We can’t say enough about Vultr. They’ve been one of our key resources for development and testing of new releases of Incredible PBX for many years. Historically, they’ve supported our open source projects through generous referral revenue although that does not apply with this special offer. If you’ve always wondered whether cloud hosting was a viable alternative to on-premise solutions, now’s your chance to kick the tires at zero cost. And the other good news is you have your choice of the following Incredible PBX offerings. Simply load the required OS or upload the ISO for the platform of your choice and follow the linked tutorials below. Enjoy!

Originally published: Friday, May 25, 2018


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



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


 

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…

Creating an OBi200 Google Voice Trunk to Use with Asterisk


Since Asterisk® will no longer be able to "talk" to Google Voice after June 17, we promised to hold our nose and document how to salvage your Google Voice trunks. Our exercise for today is to show you how to deploy an OBi 200-series device which can speak the new Google Voice language and use it as a traditional SIP bridge between Google Voice’s proprietary SIP platform and your Asterisk server. We will skip the editorializing on why Google is making a terrible mistake by discarding XMPP and forcing users to a proprietary solution necessitating a hardware purchase without first offering an open standards solution as Google’s Community Manager promised here. Promises, of course, don’t keep your phones ringing. For the whole story, see our article from last Saturday. For today, you’ll need to shell out $50 for an OBi 200 device. Once you have it in hand, feel free to read on and we’ll get you back in business. For security reasons, it should be noted that today’s setup assumes you are running an Incredible PBX® server and OBi device locally behind a NAT-based router. This will work equally well with the Incredible PBX-enhanced versions of Issabel and VitalPBX. We’ll leave it to the FreePBX® folks to figure out a solution for their proprietary distro.

Everything we’re covering below will work just as well using any of the OBi 200-series devices. We’ve simply chosen to use an OBi202 in our examples today because it supports an extra phone port. But an OBi200 works just as well if you will only be deploying Google Voice trunks (up to 3 and perhaps more) for your PBX. They retail for approximately $50 and are readily available at Amazon through the link in the right column which also provides a few shekels for Nerd Vittles to keep the lights on. As mentioned last week, Obihai crippled the OBi 110-series devices which will no longer work with the new Google Voice setup. Such a fine company that we once praised for producing our Device of the Year. And don’t worry. If you ever visit their forum, you can expect a cheery reception from the Obihai forum moderator. Here’s the response we got1 when raising concerns about the demise of Google Voice XMPP:



Registering Your OBi2x Device with OBiTALK

A Quick Start Guide accompanies your OBi hardware. Following along in the tutorial will get your OBi set up using a free (so far) OBiTALK account. When you get to Step 5, you’ll be ready to set up your Google Voice account by clicking the Google Voice Set-Up button.

Before you begin the Google Voice setup, we strongly recommend that you plug a POTS phone into your OBi device and dial ***6 to update your firmware to the latest release. Depending upon where you purchased your device, it may or may not have the latest firmware which is required to communicate with Google Voice on or after June 17.

We also recommend that you dial ***1 and obtain the DHCP-assigned IP address for your OBi. You’ll need this in a few minutes. And, while you’re at it, be sure to set the OBi up behind a NAT-based router to protect it from intrusion. Once someone gains access to your OBi, they’ve essentially got the keys to your telecom castle. So always deploy an OBi behind a hardware-based firewall that is on the same private LAN as your Asterisk PBX. Finally, on your router, be sure to reserve the DHCP-assigned IP address of your OBi for permanent use by the OBi hardware. Otherwise, the IP address of your OBi may change, and this will break the SIP gateway connection to your Asterisk server.

Finally, a word about the new OBi setup. All of your settings are now stored and managed in the OBiTALK cloud. Obihai then pushes the configuration to your OBi device. To put it charitably, this usually works but sometimes it doesn’t, and you end up with a quirky OBi setup that looks correct in the cloud but simply doesn’t work. We’ve found the simplest solution is to unplug the device and then restart it. Then check all of your cloud-based settings when the OBi device comes back to life to be sure none of your settings disappeared. Sometimes they do! In the old days, you had the option of configuring your OBi device locally; however, Obihai (now Polycom) has disabled that functionality with the new Google Voice setup presumably to disguise what they are doing under the covers to connect to Google.

Configuring a Google Voice Trunk on OBi200

To give credit where credit is due, configuring a Google Voice trunk on the OBi 200-series devices is dead simple. Login into your OBiTALK account, click on your OBi device, and then click the Google Voice Set-Up button.

Enter your Google Voice credentials when prompted, give Obihai permission to control your Google Voice account, and you’re done. Within a few seconds, the connections dialog box should show Google Voice connected on service provider SP1.

If you haven’t already done so, plug a POTS phone into your OBi device and place a call to somebody by dialing a 10-digit number. Then use another phone and call the Google Voice number you assigned to your OBi device. The POTS phone should ring. Don’t continue until you get these calls working in both directions. You’d be wasting your time.

Now we need to adjust the destination for incoming calls to your OBi device and redirect them from the POTS phone to the SP3 trunk we’ll be using to connect to your Asterisk server. We’ll leave SP2 unoccupied in case you wish to add another Google Voice trunk down the road.

To make this change, click the OBi Expert Configuration button at the bottom of the Device Configuration window. Then click OK to confirm that you know what you’re doing. Next click the Enter OBi Expert button at the top of the next form. In the left column, click Voice Services and then SP1 Service. The fifth parameter is called X_InboundCallRoute. Beside it, uncheck both the OBiTALK Settings and Device Default checkboxes. Now enter sp3(6781234567) in the Value field for X_InboundCallRoute where 6781234567 is your actual Google Voice phone number (DID). Scroll to the bottom and click the Submit button.

Finally, at the top of the left column of the form, click Return to OBi Dashboard.

Configuring OBi SIP Trunk for Asterisk

1. Login to your OBi Dashboard using a web browser . After signing up for an account and registering your OBi device, click on the OBi 200 device in the My OBi Devices list.

2. In the Device Configuration dialog, click OBi Expert Configuration button. When prompted whether you’re sure, click OK.

3. In the OBi Expert Configuration Menu, click Enter OBi Expert button.

4. In the Production Information (left) column, click Service Providers.

5. In the Service Providers listing, click ITSP Profile C General.

6. For each of these fields, uncheck OBiTALK Settings and then uncheck Device Default:

  • General:Name
  • Service Provider Info:Name
  • Service Provider Info:URL

7. Fill in the ! field Values as shown below using the private IP address of your PBX:



8. Click Submit button after checking your entries carefully.

9. In the Service Providers listing on the left, click ITSP Profile C SIP.

10. In the ITSP Profile, enter the private IP address of your PBX in the Proxy Server, Registrar Server, and Outbound Proxy fields after first unchecking both the OBiTALK Settings and Device Default checkboxes.

11. Scroll down the form to X_SpoofCallerID and uncheck both the OBiTALK Settings and Device Default checkboxes. Then check the Value field for X_SpoofCallerID.

12. Scroll down the form to X_DiscoverPublicAddress and uncheck both the OBiTALK Settings and Device Default checkboxes. Then uncheck the Value field for X_Discover PublicAddress.

13. Click Submit button after checking your entries beside the 5 red exclamation points.

14. In the Production Information (left) column, click Voice Services

15. In the Voice Services listing on the left, click SP3 Service.

16. In the SP3 Service Profile, fill in the 5 fields in which the OBiTALK Settings checkbox is unchecked. The AuthUsername and AuthPassword entries will be used to authenticate to your PBX so be sure to choose a very secure password. It’s your phone bill. The URI field actually makes the trunk connection to your PBX so replace the 192.168.0.82 entry shown with the actual IP address of your PBX.

17. In the SIP Credentials section of the form, make certain that X_EnforceRequestUserID is unchecked. If not, uncheck both the OBiTALK Settings and Device Default checkboxes and then uncheck X_EnforceRequestUserID.

18. If you do not want to pass the CallerID number with your calls, in the Calling Features section of the form, be sure to check AnonymousCallEnable after unchecking both the OBiTALK Settings and Device Default checkboxes.

19. In the Service Providers listing on the left, click ITSP Profile A SIP.

20. Be sure X_SpoofCallerID is checked.

21. Click Submit button after checking your entries carefully.

Configuring Incredible PBX GUI for an OBi200

On the Incredible PBX side, log into the GUI using a web browser. We’ll be adding a SIP trunk, an outbound route, and an inbound route to process calls to and from the OBi device.

Add a SIP Trunk with a Trunk Name matching whatever you used in your OBi SIP credentials, e.g. obi200 or obi202. Plug in your Outbound CallerID to match your Google Voice phone number. In the Dialed Number Manipulation Rules tab, add a Match Pattern of NXXNXXXXXX. In the SIP Settings tab for Outgoing, the Trunk Name should match whatever you used on the OBi side, e.g. obi200 or obi202. In the PEER DETAILS, enter the following using the default username and password you assigned on the OBi side. Normally, port 5061 is the default port assigned on the OBi side. If you get a failed registration, try 5060 and then 5062 and 5063. Click Submit and reload your dialplan when finished.

type=friend
defaultuser=obi200
secret=your-password
qualify=yes
port=5061
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite

For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, your Incredible PBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. When your Outbound Route setup looks like the following, click Submit and reload your dialplan.



For Inbound Call Routing, create an Inbound Route specifying a DID Number to match your Google Voice number. Choose a Call Destination to meet your own requirements, e.g. an extension, ring group, or IVR. Then click Submit and reload your dialplan.

Now you’re ready to test an outgoing call by dialing the OBi prefix (624) plus a 10-digit number. Then place a call to your Google Voice number using your cellphone and be sure Asterisk routes it to the destination you specified in your inbound route above.

Configuring VitalPBX to Use an OBi200

Truth be told, we weren’t bright enough to figure out how to configure the VitalPBX Trunk using credentials so we simply set up the SIP trunk using IP address authentication with the IP address of the OBi device. It works just as well and just goes to prove there’s always more than one way to skin a cat. So here’s the Trunk configuration on the VitalPBX side. The only entry you will need to change is the Host IP address for your OBi device. If you don’t know it, plug a phone into the OBi and dial ***1.

NOTE: For the Username and Description fields below, be sure to match what you used on the OBi side (above) for your SIP credentials, i.e. obi200 or obi202. If they don’t match on both devices, you won’t get a successful connection. Our apologies for mixing apples and oranges in the screenshots.



For Outbound Call Routing, we recommend an Outbound Route using the 624 (OBI) prefix and 10-digit numbers. For example, if a user dials 624-888-1234567, the VitalPBX server would place a call using the OBi’s Google Voice trunk to 1-888-1234567. Here’s the Outbound Route setup to make that happen:



For Inbound Call Routing, go to PBX:External:Inbound Routes and add an inbound route and destination for calls from your 10-digit Google Voice number. Or you can use the Default Inbound Route which we explained in our previous VitalPBX tutorial. Basically, you set up an Inbound Route with a Description and Routing Method of Default. All the other fields should be left as is except for the Inbound Destination. For the destination, you can choose an IVR, Extension, Ring Group, etc. to meet your own requirements.

Originally published: Monday, May 14, 2018


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.


SPECIAL TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID. Enjoy!



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.
 



  1. You can always find a little humor in insults if you dig deep enough. Ironically and unbeknownst to our pal, Steve, it was Sherman Scholten and his OBi development team that were among the first Google Voice "freeloaders." Only years later after Google Voice was integrated into FreeSwitch did Josh Culp at Digium perfect a clean way to integrate Google Voice into the Asterisk platform. []

Incredible PBX in the Cloud: A $10/Year VoIP Cloud Platform

We’ve been inching toward a new low-cost plateau for VoIP cloud providers, and today we have a new milestone that finally makes running VoIP servers out of your home or office look like the horse-and-buggy days. $10 a year now buys you a cloud platform that is less expensive than the cost of electricity to run a server on premise. You get 2GB of RAM, 20GB of SSD storage, two virtual core processors, and 2TB of monthly bandwidth. If you prepay for 3 years, you can double either the RAM or SSD storage by simply opening a ticket after you sign up. It’s a near perfect platform for Incredible PBX 13-13 with CentOS 6.9. Add a Google Voice trunk and you get unlimited calling in the U.S. and Canada combined with a feature set that you’ll be hard-pressed to find on any PBX at any price. Putting all the pieces in place is about as simple as preparing slice-and-bake cookies, and you’ll be up and running before the cookies come out of the oven. Skip that hamburger lunch and come join the VoIP revolution!



So what’s the catch? Well, there’s no catch with Incredible PBX 13-13 and CentOS 6.9. But this HiFormance platform uses OpenVZ with SolusVM, and SolusVM has some serious bugs with their CentOS 7 and Debian 9 implementations. That rules out using VitalPBX, Issabel, or Wazo. Someone always asks, "If the platform is so great, why aren’t you using it?" And our answer is we are. We have deployed HiFormance cloud-based VoIP servers running Incredible PBX 13-13 in Atlanta, Buffalo, Chicago, and Los Angeles without any hiccups in service. Performance is excellent. Support is excellent. So run, don’t walk, to sign up for one of these before they’re all gone. You won’t be disappointed. Just fill out the entries as shown above once you log into the HiFormance site. Nerd Vittles receives no commissions from signups.

Getting Started with Incredible PBX 13-13

Once your virtual machine is up and running with CentOS 6.9, log into your server as root and issue the following commands to get started. Use the first command to immediately change your root password. Then you’ll be ready to begin the Incredible PBX install. It’s a two-step process. First, the installer will bring your version of CentOS up to current specs and load the necessary packages to support Asterisk® and FreePBX®. The first stage takes 22 minutes.

passwd
cd /root
yum -y update
yum -y install net-tools nano wget tar
wget http://incrediblepbx.com/incrediblepbx-13-13-LEAN.tar.gz
tar zxvf incrediblepbx-13-13-LEAN.tar.gz
rm -f incrediblepbx-13-13-LEAN.tar.gz
./IncrediblePBX-13-13.sh

When the base install finishes, your server will reboot. Simply log back in as root and run the installer a second time. Be sure your console window is at least 80 x 28, or the install will fail. If in doubt, expand it to full screen. You’ll be prompted whether to implement Google Voice plain text or OAuth 2 passwords.1 OAuth is strongly recommended. In fact, OAuth is required if you wish to install the Whole Enchilada upgrade which gets you several dozen preconfigured applications for Asterisk. Make your selection, and the installer will work its magic. Return in 12 minutes.

./IncrediblePBX-13-13.sh

Reboot one final time when the installer finishes the setup, and Incredible PBX LEAN will be ready to go. Log back in as root. This will kick off the Automatic Update Utility to load any last minute additions, bug fixes, and security patches. After the status menu displays, run the following apps to set a very secure admin password for web access to the GUI and to choose your default time zone:

/root/admin-pw-change
/root/timezone-setup

One of the unique features of Incredible PBX 13-13 is that most of the major components of the aggregation including Asterisk are compiled from source code on the fly. This has several advantages. First, you always get the latest version of the source code. And, second, the source code is available on your server so that you can make any future modifications desired to meet your own unique requirements. You won’t find this in any other VoIP implementation. It’s one of the reasons Incredible PBX takes a bit longer to install than many of the canned offerings that rely upon precompiled packages that are difficult to modify.

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration.

Using the Incredible PBX 13-13 Web GUI

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it. You can safely ignore the warning about a missing swap file. You have plenty of RAM, and OpenVZ platforms don’t permit swap files. If you’re worried about it, choose the 3-year prepayment option and double your ram from 2GB to 4GB which is more than ample for even the busiest PBXs.

NOTE: If you plan to upgrade to the Whole Enchilada, you can skip the rest of this section. It’s for those that wish to roll their own PBX from the ground up.

To get a basic system set up so that you can make and receive calls, you’ll need to add a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You’ll also need a SIP phone or softphone to use as an extension on your PBX. Our previous tutorial will walk you through this setup procedure. Over the years, we’ve built a number of command line utilities including a script to preconfigure SIP trunks for more than a dozen providers in seconds. You’ll find links to all of them here.

Continue Reading: Configuring Extensions, Trunks & Routes

Reconfiguring PortKnocker for OpenVZ

By default, PortKnocker monitors activity on eth0. Most OpenVZ platforms including HiFormance use venet0:0 as the default Ethernet port. Issue the following commands to get PortKnocker up and running. Then pbxstatus should show PortKnocker working.

echo 'OPTIONS="-i venet0:0"' >> /etc/sysconfig/knockd
service knockd restart
pbxstatus

Reconfiguring NeoRouter VPN for OpenVZ

On OpenVZ platforms including HiFormance, you’ll need to enable TUN/TAP in the Control Panel for your VPS. After adjusting the setting, reboot your server. Then the NeoRouter VPN client will function properly.

Upgrading to Incredible PBX Whole Enchilada

There now are two more pieces to put in place. The sequence matters! Be sure to upgrade to the Whole Enchilada before you install Incredible Fax. If you perform the steps backwards, you may irreparably damage your fax setup by overwriting parts of it.

The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Be advised that this upgrade will overwrite all of your existing Incredible PBX setup including any extensions, trunks, and routes you may have created previously. You also will be prompted to reset all of your passwords as part of the upgrade. Install time: 2 minutes.

cd /root
./Enchilada*

If you accidentally installed Incredible Fax before upgrading to the Whole Enchilada, you may be able to recover your Incredible Fax setup by executing the following commands. It’s worth a try anyway.

amportal a ma install avantfax
amportal a r

Installing Incredible Fax with HylaFax/AvantFax

You don’t need to upgrade to the Whole Enchilada in order to use Incredible Fax; however, you may forfeit the opportunity to later upgrade to the Whole Enchilada if you install Incredible Fax first. But the choice is completely up to you. To install Incredible Fax, log into your server as root and issue the following commands. Install time: 2 minutes.

cd /root
./incrediblefax13.sh

After entering your email address to receive incoming faxes, you’ll be prompted about two dozen times to choose options as part of the install. Simple press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password; however, you first will be prompted for your Apache admin credentials which were set when you installed Incredible PBX 13-13 LEAN or the Whole Enchilada. Then you’ll be asked to change your AvantFax password.




Upgrading to IBM Speech Engines

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. While IBM’s services are not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free STT voice recognition services. And the first million characters of text-to-speech synthesis are FREE every month as well. So let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada. Here’s our tutorial that will walk you through the one-time IBM setup.

Next, login to your Incredible PBX server and issue these commands to update your Asterisk dialplan and edit ibmtts.php:

cd /var/lib/asterisk/agi-bin
./install-ibmtts-dialplan.sh
nano -w ibmtts.php

Insert your credentials in $IBM_username and $IBM_password. Verify that $IBM_url matches the entry provided when you registered with IBM. Then save the file: Ctrl-X, Y, then ENTER. Now reload the Asterisk dialplan: asterisk -rx "dialplan reload". Try things out by dialing 951 (news) or 947 (Weather) from an extension registered on your PBX.

To get IBM’s Speech to Text service configured, while still logged in to your Incredible PBX server, issue these commands to edit getnumber.sh:

cd /var/lib/asterisk/agi-bin
nano -w getnumber.sh

Insert your API_USERNAME and API_PASSWORD in the fields provided. Then save the file: Ctrl-X, Y, then ENTER. Update your Voice Dialer (411) to use the new IBM STT service:

sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf
sed -i '/\\[from-internal-custom\]/r ibm-411.txt' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines." Check back next week for the Whole Enchilada apps tutorial.

Configuring Google Voice with Incredible PBX

The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can’t beat the price, and it has worked reliably for many, many years. There are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.

Here are the initial setup steps on the Google side:

1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You’ll need to provide a U.S. phone number to verify your account by either text message or phone call.



2. Once you have verified your account by entering your verification code, you’ll get a welcome message from Mr. Google. Click Continue to Google Voice.



3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.



4. Once your phone number has been verified, choose a DID in the area code of your choice.



Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here’s a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google’s Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC’s browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there’s still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it’s labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.

5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller’s Number, and Global Spam Filtering:checked. The remaining entries should be blank.

6. Google Voice configuration is now complete. Sign out of your Google Voice account.


The Simonics GV-SIP Gateway Solution. Here’s the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we’ll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we’ll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.

Now you’re ready to set up an account on the Simonics site. With this Nerd Vittles link, there’s a one-time fee of $4.99.

1. Start by registering your new Google account.

2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.

3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.



4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.

Now it’s time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:

1. Untick the Disable Trunk check box.

2. In Outbound CallerID, insert your 10-digit Google Voice number.

3. In username, insert GV1 followed by your 10-digit Google Voice number.

4. In secret, insert your Simonics SIP password.

5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)

6. In the Registration String after the colon, insert your Simonics SIP password.

7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.

8. Click Submit Changes and then Reload the Dialplan when prompted.


Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.

1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.

2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.

3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).

  3a. Check the box: Use your own OAuth credentials
  3b. Enter Incredible PBX OAuth Client ID:

466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com

  3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
  3d. Click Close

4. Click Step 1: Select and Authorize APIs (as shown below)

  4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
  4b. Click Authorize APIs (blue) button.

5. Click Step 2: Exchange authorization code for tokens

  5a. Click Exchange authorization code for tokens (blue) button

  5b. When the tokens have been generated, Step 2 will close.

6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn’t permanently store it!

7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.

Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.

Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don’t skip any. It’s easy once you get the hang of it. And you’ll sleep better.

Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.

If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, 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.

IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart

Incredible PBX Wholesale Providers Access

Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.

Continue Reading: Configuring Extensions, Trunks & Routes

Don’t Miss: Incredible PBX Application User’s Guide covering the 31 Whole Enchilada apps

Originally published: Monday, April 16, 2018


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.


NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!



Need help with Asterisk? Join our new MeWe Support Site.


 

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. Complete Google Voice setup tutorial is available here. []

Meet the New Kid on the Block: Introducing (free) VitalPBX




If you liked Ombutel, you’re going to love VitalPBX. If you’ve never heard of Ombutel but you live and breathe Asterisk®, you’re still going to love VitalPBX. For everyone else, you’re going to love VitalPBX. In addition to an impressive collection of commercial modules, this month’s release of the VitalPBX 2.0 Unified Communications Platform provides the slickest user interface in the VoIP universe. It includes new support for PJsip, DPMA and Digium phones, XMPP chat, video conferencing, WebRTC, and our favorite, Custom Contexts. What began several years ago as a joint development project between Telesoft and Xorcom is now an independent venture of Telesoft. If you love Features, VitalPBX has no equal:


VitalPBX has many open source and GPL components including Asterisk 13.19.0, however, VitalPBX is a freeware product much like FreePBX® which blends commercial modules and proprietary components into its distribution. It’s not our favorite business model, but we certainly understand the rationale given the disappointing GPL history in the VoIP space. For our testing purposes, Telesoft has generously provided free licenses to commercial modules. We would hasten to add that no features requiring payment were used in this article or in the demo applications accompanying it. We will cover the commercial applications requiring payment at a later date.

Incidentally, when you get around to exploring the commercial offerings, keep in mind that all of them come with a free tier to let you try things out:

  • Custom Contexts – 1 free context
  • IVR Stats – 1 free IVR
  • Sonata Switchboard – 1 free layout for 15 extensions
  • Sonata Billing – free for 8 extensions
  • Sonata Recordings – free for 8 extensions
  • Domotic – completely free
  • Phone Books – completely free
  • Bulk Extensions – completely free

Today we want to walk you through getting a VitalPBX server set up so that you can kick the tires for yourself. Down the road, we’ll demonstrate the ease with which you can add your own components including Incredible PBX® to the mix. If you are accustomed to setting up FreePBX-based Asterisk servers, then installation and configuration of VitalPBX will be a walk in the park. Currently, you install VitalPBX from an ISO so you have a choice of platforms: dedicated hardware, VMware ESXi, VirtualBox, or a limited number of cloud platforms such as Vultr that support custom ISO installs. Be sure to read our security warnings below before choosing a cloud-based platform without a hardware-based firewall.

A Word About Security. VitalPBX includes both an IPtables firewall configurator for firewalld and a Fail2Ban intrusion detection setup that is impressive. Having said that, the IPtables firewall is activated but allows unrestricted SIP and web access with no rules to thwart SipVicious-style attacks. Unless you’re an expert in firewall design, we strongly recommend deployment of VitalPBX on a private LAN behind a hardware-based firewall or home router with no port forwarding. That will block intrusion attempts without encountering NAT problems which VitalPBX and Asterisk 13 now handle with ease.

Getting Started. Begin by downloading the VitalPBX 2.0 ISO to your desktop. The ISO installation process is a traditional CentOS® 7 procedure so you can follow one of our existing VoIP tutorials to get things set up on the platform of your choice. Once the install finishes, use a web browser to access the IP address of your VitalPBX server. You’ll be prompted to set up an admin password for GUI access and then you register your server with Telesoft. Should you ever forget your admin password, here’s how to force a reset on your next login from a browser:

mysql ombutel -e 'update ombu_settings set value = "yes" where name = "reset_pwd"'

After logging in, you’ll be presented with the VitalPBX Dashboard:





Navigation Tips. The GUI is incredibly intuitive, but there’s always a learning curve with something new. We’ll save you a little stumbling around looking for things or wondering why your settings in the UI didn’t take. Here’s a quick cheat sheet. All of the UI features are housed under menus in the left column. When you choose an option, it opens a submenu. And, when you click + beside an item on the submenu, it exposes additional choices. For example, to work on Outbound Routes, you’d choose PBX, External +, Outbound Routes:


Two other important icons are housed in the upper right corner of the GUI. Whenever you add or make changes to settings in the GUI, you need to reload the Asterisk dialplan by clicking on (1) the flashing icon. Otherwise, your settings will not be available. Ask us how we know. 🙂

After you add a new extension, trunk, or route, you’ll see (2) the four-bar icon which you click to access existing settings which you’ve already entered. Otherwise, you’ll be staring at a blank screen without your new entries. There’s nothing more disconcerting than adding a few extensions only to have them disappear the next time you navigate to PBX:Extensions. 🙂



Finally, at the top of the center panel of the GUI, VitalPBX (literally) keeps tabs on items you’ve recently worked on. It makes it extremely convenient to return to the item without having to once again drill down through the menus:



Initial Setup. As with most PBXs, the initial setup involves creating some Extensions, connecting some Trunks, and setting up Outbound and Inbound Routes to process calls to and from your PBX. The other hundreds of features are pure gravy which you can explore at your leisure. If we covered them all, you’d be reading a book instead of an article.

Extension Setup looks like this using VitalPBX to generate the extension password:



Trunk Setup. You can use Google Voice with the Simonics GV/SIP Gateway for free calling in the U.S. and Canada. There is a one-time setup charge of $4.99 if you follow this Nerd Vittles link. We recommend using Google Voice for outbound calls where possible. Then, for inbound calls and redundancy, add a separate trunk with a customized DID from a provider such as our platinum sponsor, Vitelity. See the end of this article for a deal you can’t refuse. The VitalPBX Trunk setup in PBX:External:Trunks:SIP would look like the following for the Simonics GV/SIP gateway:



Outbound Route Setup is virtually identical to the FreePBX format. Here’s a typical Google Voice route to let users dial 10-digit numbers while letting Google discard expensive NANPA calls to problematic area codes in the Caribbean and elsewhere. We actually recommend adding a second Dial Pattern for 1NXXNXXXXXX so that calls dialed with both 10 and 11-digits are supported. This will also facilitate implementation of some of the Incredible PBX add-ons down the road.



Inbound Route Setup also is similar to FreePBX. A default route can be configured by simply defining the Route Description as Default and specifying a Destination for all incoming calls that don’t otherwise have a matching inbound route.

Email Configuration. One of the other things you’ll want to get working is email delivery for Voicemails. The VitalPBX solution is the best in the business. It supports Gmail as a RelayHost out of the box. For residential users where your ISP blocks downstream SMTP mail servers, this is a godsend. Setup couldn’t be easier. Navigate to Admin:System Settings:Email Settings. For Server, click Use External Mail Server. For Provider, click Gmail and enter your full Gmail account name and password. Click Save and Reload your Dialplan. Then send yourself a test message by entering an email address and clicking the Envelope icon.

Updating Time Zone. If the date command incorrectly displays the time on your server, you can change it with the following commands using your correct zone in the second command:

timedatectl list-timezones
timedatectl set-timezone America/New_York

What’s Next? You now have a perfectly functioning PBX. Connect one or more softphones or SIP phones, and you’re ready to go. As we mentioned at the outset, the next step is to explore all of the menu options and review the VitalPBX Reference Guide. It really is a book!

The Fun Stuff. The icing on the VitalPBX cake is the add-on applications. Some are free, some are limited in some way, and some are commercial. You can review what’s available here. Then load the currently available listing into the GUI by choosing Admin:Add-ons:Add-ons:Check Online. To get started, install Bulk Extensions (free), Custom Contexts (one free context or $50 for unlimited), and Phone Books (free). Once you’ve installed all three, refresh your browser and go to PBX:Applications:Custom Contexts.

Step #1. Set up a Custom Context like this. Then click Save/Update and Reload Dialplan.



Step #2. Adjust Destination of Inbound Route to point to Incredible PBX Custom Context.

Step #3. From the Linux CLI while logged in as root, use nano to create the following file: /etc/asterisk/ombutel/extensions__80-1-incrediblepbx.conf:

[incrediblepbx]
exten => s,1,Answer
exten => s,n,NoOp(My custom context)
exten => s,n,Dial(SIP/701,30)
exten => s,n,return()

Step #4. Reload your Asterisk dialplan: asterisk -rx "dialplan reload"

Step #5. Place a call to an incoming trunk on your PBX while watching the Asterisk CLI. The tail of the incoming call should look something like the following which shows the incoming call directed to the Custom Context and from there to extension 701.



Now that you understand the VitalPBX theory behind Custom Contexts, you’ll be ready to dive into Incredible PBX applications which will be coming soon to a VitalPBX platform near you.

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

Homework. Yes. Everyone needs a little homework once in a while. Before our next chapter in the VitalPBX saga, you’re going to need an IBM Cloud account with access to Watson TTS and Watson STT. It’s free. These services will be used for the Incredible PBX TTS and Voice Recognition apps for Asterisk including News and Weather reports as well as Voice Dialing with AsteriDex. This Nerd Vittles tutorial will walk you through getting your IBM account set up. Don’t install any of the scripts in that tutorial. We’ll have fresh ones in coming weeks customized for VitalPBX. For home and SOHO use, both IBM access and our scripts are FREE.

Coming Attractions. We’ve set up a VitalPBX demo server with VMware ESXi running on our private LAN. Most of the Incredible PBX demo applications already are operational, and you’re more than welcome to try them out by calling the IVR at 1-843-606-0555. Many of these apps make use of the IBM Cloud services for voice recognition and text-to-speech content rendering so you can preview what you’ll be getting in our next VitalPBX chapter.

  • 0. Chat with Operator — connects to extension 701
  • 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
  • 2. Conferencing – log in using 1234 as the conference PIN
  • 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
  • 4. Lenny – The Telemarketer’s Worst Nightmare
  • 5. Today’s News Headlines — courtesy of Yahoo! News
  • 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
  • 7. Today in History — courtesy of OnThisDay.com
  • 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
  • 9. DISA Voice Dialer — say any 10-digit number to be connected
  • *. Current Date and Time — courtesy of VitalPBX

Continue Reading:

Introducing the Incredible PBX Custom Context for VitalPBX
VitalPBX in the Cloud: Two $6/month Providers with Backups
VitalPBX Security: Firewall, PortKnocker, & NeoRouter VPN
VitalPBX on the Desktop: Introducing VitalPBX for VirtualBox

Originally published: Monday, March 19, 2018





Need help with VitalPBX? Visit the VitalPBX 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.
 



The SMS Toolkit: Integrating Text Messaging into Asterisk

Unless you’ve been living under a rock, you probably already know that most folks spend a lot more time texting on their smartphones rather than talking. So it only made sense to develop some useful SMS tools to get the most out of your Asterisk® PBX. Today we’re pleased to introduce version 1 of an SMS Toolkit for Incredible PBX® using any SMS-enabled DID from either Vitelity or VoIP.ms. Just text a simple message to your PBX, and Incredible PBX will do the heavy lifting and either call you back with the results or reply to your text message. You can whitelist an IP address in your firewall or retrieve news headlines or a weather forecast. You can also look up a phone number in your AsteriDex phone book and place a call through your PBX using either the Voice Dialer or speed dial codes. You can enable call forwarding from your PBX extension to your smartphone, or a simple SMS command brings the calling flexibility of DISA to your smartphone. Here’s a list of supported SMS commands:

  • help – Plays a list of supported SMS commands
  • news – Retrieves latest news headlines from Yahoo
  • weather – Retrieves weather report by zip code
  • wolfram – Siri-like query to Wolfram Alpha
  • whitelist – Whitelist a new IP address in your firewall
  • disa – Use DISA calling from your smartphone with password
  • cf on – Enable call forwarding from PBX extension to smartphone
  • cf off – Disable call forwarding from PBX extension
  • cf status – Status of call forwarding on PBX extension
  • asteridex – Use AsteriDex Voice Dialer to place a call from PBX
  • odbc – Use AsteriDex speed dial codes to place a call from PBX
  • sms – Dictate an SMS message and deliver from Google Voice on PBX

Prerequisites for SMS Toolkit Deployment

To get started, you’ll need a DID from Vitelity or VoIP.ms that supports SMS messaging. You’ll be hard-pressed to beat our Vitelity DID special which is summarized at the end of this article, but the choice is all yours. The way this works is you provide a forwarding email address in the Vitelity or VoIP.ms portal for delivery of incoming SMS messages. These emails will be sent to your PBX where we will use SendMail and our mailcall script to process the messages and deliver the results. SMS messaging is a free add-on with both Vitelity and VoIP.ms DIDs unlike some other providers.

Since Vitelity or VoIP.ms will be delivering incoming SMS messages by email, it means you’ll also need a dedicated account and fully-qualified domain name (FQDN) for your server, e.g. smsuser@mypbx.mydomain.com. While a dynamic IP address will work if you implement automatic FQDN updating on your PBX, a static IP address for your PBX is obviously preferable and all of our recommended cloud solutions provide that including RentPBX, Digital Ocean, Vultr, and HiFormance.

Implementing the SMS Toolkit for Asterisk

Let’s walk through the steps to put all the pieces in place for the SMS Toolkit:

  1. Add a Dedicated Account to Linux for SMS Messaging
  2. Configure PBX for Receipt of Incoming SMS Emails
  3. Obtain and Configure DID with Vitelity or VoIP.ms
  4. Install and Configure MailCall Components

1. Adding a Dedicated Account to Linux

To simplify the task of sifting through incoming emails, we’ll want to create a new Linux user account that can be dedicated to receipt of SMS email messages. The second and third commands will verify that the account has been created with support for incoming mail. Just log into your server as root and issue the following commands:

adduser smsuser --shell=/bin/false --no-create-home --system -U
ls /var/mail/smsuser
mail -u smsuser

 

2. Configuring SendMail to Receive Inbound Email

By design, both SendMail and the Incredible PBX firewall block incoming email. We’re going to change that but, in doing so, we wish to caution that we don’t want to turn your server into an open mail relay for the spammers of the world. Once we’ve opened up your server to receive email, it’s important to test it to be sure it’s not insecure. Because the SMS Toolkit is intended to be a dedicated application just for you as administrator of your server, it’s equally important not to publicize the FQDN of your server. Once the spammers find your email address, incoming email can be just a big a problem as serving as an open mail relay.

To add a firewall rule in Incredible PBX to support incoming SMTP mail traffic, issue the following commands:

echo "iptables -A INPUT -p tcp -m tcp --dport 25 -j ACCEPT" >> /usr/local/sbin/iptables-custom
iptables-restart

Configuring SendMail to receive incoming email requires a few changes in /etc/mail/sendmail.mc followed by a restart of the SendMail service.

Using your favorite editor: nano -w /etc/mail/sendmail.mc

1a. Search for the following line:

DAEMON_OPTIONS(`Port=smtp,Addr=127.0.0.1, Name=MTA')dnl

1b. Replace that line with the following two lines:

dnl # DAEMON_OPTIONS(`Port=smtp,Addr=127.0.0.1, Name=MTA')dnl
FEATURE(`dnsbl',`dnsbl.njabl.org',`"550 Mail from " $&{client_addr} " rejected"')dnl


2a. Search for the following line:

FEATURE(`accept_unresolvable_domains')dnl

2b. Replace the line with the following and save the file:

dnl # FEATURE(`accept_unresolvable_domains')dnl

 
3. If you need an FQDN or dynamic DNS support, you can’t beat free. Here are some options.

4. Issue the following commands to complete the setup and restart SendMail:

yum -y install sendmail-cf
cd /etc/mail
make
echo "127.0.0.1 insert-your-server-FQDN-here" >> /etc/hosts
service sendmail restart

 
5. Using a browser from your desktop PC, test your server’s IP address to make certain it is not an open mail relay. All tests should report "Relaying Denied" or "Invalid route address."

6. Test your new mail account by sending an email to smsuser@your-server’s-fqdn. Wait a bit and check for email: mail -u smsuser. Then delete the email: > /var/mail/smsuser

3a. Configuring Vitelity DID for SMS Email Relay

With Vitelity DIDs, the first step is to order a DID that supports SMS, most do. Next, you need to decide whether this DID will be used for other purposes, such as serving as a trunk on your PBX for receipt of incoming calls. If this is your plan, then review the Vitelity offer at the bottom of this page. You’ll be hard-pressed to beat the price or the feature set. And you’ll also be providing support for our open source projects. Vitelity has been a Platinum Sponsor of Nerd Vittles for almost a decade.

If you only want to use the DID to support SMS messaging, then there’s little reason to sign up for the unlimited calling plan. Instead, choose the pay-per-minute (PPM) plan for your DID. It costs $1.49 a month. Don’t even both registering the trunk which will save your having to pay for misdialed calls and spam. SMS messages are free.

Once your DID is set up, go to My Numbers -> Local in the Vitelity web portal and choose SMS from the Action pull-down menu of your new DID:

In the SMS dialog, set up a password for messaging, disable international messages, and enter the email forwarding address for your incoming SMS messages. Save your settings, and you’re good to go.

3b. Configuring VoIP.ms DID for SMS Email Relay

If you plan to dedicate a DID to SMS messaging, two advantages of the VoIP.ms offering are (1) the price ($0.85/month on the pay by the minute plan with $0.40 setup fee and (2) you can forward incoming SMS messages to another SMS number (such as your smartphone) in addition to an email address if you want to use the DID for traditional SMS messaging while also deploying our SMS Toolkit. As noted above, there’s no charge for SMS messages at this time although VoIP.ms has warned (for years) that they may begin charging a penny a message. As long as Vitelity is free, I wouldn’t worry. 🙂

To get started, sign up for a VoIP.ms account and order a DID with SMS support. A cellphone is displayed beside each DID that supports SMS in their ordering page. As with Vitelity, there’s no need to register the trunk on your PBX if you only plan to use the SMS messaging component. Once your DID is provisioned, choose DID Numbers -> Manage DIDs in the VoIP.ms portal. Then edit the DID you just purchased. At the bottom of the form, fill in the SMS section as shown below:

4. Installing and Configuring MailCall on your PBX

MailCall was specifically designed for Incredible PBX 13-13 Full Enchilada but should work without modification with the latest version of Incredible PBX for Issabel 4. For other platforms including Debian, Ubuntu, and Raspbian, some adjustments may be necessary, and we have not yet had time to work out the kinks. There will be some so please hold off. Just to restate the obvious. Many of the MailCall features will not work until they are first configured on your server, e.g. Voice Dialing, Wolfram Alpha, and Voice SMS Messaging. All of the setups are covered in the Full Enchilada tutorial.

After logging into your server as root, issue the following commands to install MailCall:

cd /
wget http://incrediblepbx.com/MailCall.tar.gz
tar zxvf MailCall.tar.gz
rm -f MailCall.tar.gz

 

For the DISA and Firewall WhiteList components of MailCall, a 5-digit PIN is required for obvious reasons. This needs to be set in two places: (1) in two chunks of dialplan code (/tmp/sms-dialplan.txt) to be added and (2) at the top of the mailcall script itself (/root/mailcall). Just search for XXXXX and replace the five X’s with a 5-digit secure PIN in all three places. Then issue the remainder of the commands below:

nano -w /tmp/sms-dialplan.txt
nano -w /root/mailcall
cat /tmp/sms-dialplan.txt >> /etc/asterisk/extensions_custom.conf
echo "asterisk ALL = NOPASSWD: /sbin/iptables" >> /etc/sudoers
echo "*/2 * * * * root /root/mailcall > /dev/null 2>&1" >> /etc/crontab
asterisk -rx "dialplan reload"

 

As a security precaution, you can only use SMS Toolkit to forward and unforward calls to your cellphone from a PBX extension designated for your use. You can associate more than one cellphone with a given extension, but you can’t associate multiple extensions with a single cellphone. To set up the association between your cellphone and an extension on your PBX, issue the following command while logged in as root where 8431234567 is your cell phone number and 701 is the associated extension:

asterisk -rx "database put CELL 8431234567 701"

 

Sending an SMS message of CF ON to your DID from 8431234567 will automatically forward extension 701 calls to your cell. Sending CF OFF will disable call forwarding for extension 701. Sending CF STATUS will retrieve the current status of call forwarding for extension 701.

Finally, a few words about the SMS whitelist command. It can be used in two ways. If you just text whitelist, then you will get a call back that first prompts for your PIN. You then will be prompted for the IP address to whitelist. Using your cellphone, enter the IP address using * for periods, e.g. 1*2*3*4 becomes 1.2.3.4. The alternative whitelist option doesn’t require a callback. Just send a text message with whitelist pin ip-address using periods, not *, e.g. WHITELIST 98765 1.2.3.4 would whitelist 1.2.3.4 if your PIN was correctly entered as 98765 and matches the entry in /root/mailcall. Enjoy!

Published: Monday, February 26, 2018



NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID.

Why not add 300 New Wholesale Providers to Make Asterisk Shine while visiting TelecomsXchange. Enjoy!



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…

300 New Wholesale Providers Make Asterisk Shine


For many years, we’ve offered Vitelity’s $3.99 Unlimited DID special with a DID, 4 channels, unlimited inbound calling, and free text messaging. It’s a deal no VoIP user should pass up. And today we have another revolutionary development for Asterisk® deployments. Once in a while we feature a carrier with great calling rates. Today we’re introducing a service providing access to 300+ wholesale carriers, all under one roof. Almost 30 BILLION billed minutes already have been logged through TCXC so you’re in good hands!1 You can choose from any or all of their 300 wholesale VoIP carriers worldwide to make outbound calls at the same wholesale rates offered to the very largest resellers. Simply pay a 13% surcharge in lieu of the $650 annual fee, and TelecomsXchange (TCXC) will provide you access to their entire suite of wholesale carriers together with state-of-the-art tools to manage all of the services.2 You’ll never have to haggle with individual carriers or provide funds on a piecemeal basis to use any of the services. TCXC already has done the negotiating for you and TCXC handles financial reimbursements to carriers based upon the services you use. There’s more good news. When compared to commercial providers, TCXC’s one-second billing increment on most routes will recoup a healthy portion of the 13% wholesale surcharge. Here are a few sample per minute wholesale rates (all with one-second billing) to whet your appetite:

  • $.0000 – U.S. TollFree
  • $.0009 – U.S.
  • $.0010 – Cyprus
  • $.0011 – Canada
  • $.0019 – Germany
  • $.0021 – U.K. (London)
  • $.0042 – China

What does a penny buy? 11-minute call to U.S., 10-minute call to Cyprus, 9-minute call to Canada, 5-minute call to Germany, 5 minute call to England, or 2½-minute call to China.

 
If you’re new to wholesale terminations, be advised that carriers change their rates regularly and, from time to time, every carrier experiences outages. Not to worry. For a modest additional charge, TelecomsXchange will manage rates and provide automatic failover for carrier outages. Simply choose TelecomsXchange as your preferred provider to the outbound destinations desired.

Before we get into the nuts and bolts of configuring Asterisk to use TCXC carriers for wholesale call terminations, let’s spend a minute discussing the architecture of the FreePBX® trunk and outbound routes model. In this design which you will find in most Incredible PBX® implementations including Issabel 4, Incredible PBX for CentOS and Ubuntu, and Incredible PBX for the Raspberry Pi as well as in other Asterisk distributions including AsteriskNOW® and the FreePBX Distro®, the administrator specifies Trunks for each provider and then assigns Outbound Routes for calls using those providers. When calls are placed, FreePBX chooses an Outbound Route based upon the dial string match specified in the route. If you have a dozen outbound routes, dialed numbers are analyzed against dial strings specified in each Outbound Route, and the routes are examined from the top to the bottom of the list. Once FreePBX chooses an Outbound Route to process a call, that ends the Outbound Route selection process. No other Outbound Route is ever considered whether it has a matching dial string or not. And it doesn’t matter whether the call fails or not, no other Outbound Route is attempted. The good news is that, within every Outbound Route, you can specify multiple Trunks which will be used in the order you’ve chosen to complete the call. If the ninth trunk happens to be the first trunk that doesn’t experience congestion, then the call will be routed to carrier #9. Keep in mind that calls to the previous eight carriers have to be attempted before we ever get to carrier #9. For this reason, it is important to create a Trunk for every carrier and specify multiple Trunks in every Outbound Route to avoid failed calls. Or, as noted above, you can specify TelecomsXchange as your final Trunk in every Outbound Route and leave it to TCXC to identify a working carrier to complete your call. In this way, you never have to worry about failed calls even though some may cost a little more depending upon carrier outages. So that’s how VoIP terminations work. You’re now an expert!

Getting Started with TelecomsXchange

The first step in your wholesale VoIP adventure is to sign up for an account with TelecomsXchange. Unless you’re chomping at the bit to pay the $650 annual fee, use our referral link. Your PBX will need a public IP address but, if it happens to be a dynamic IP address assigned by your provider, don’t worry. It’s easy to change it down the road, and we’ll show you how. Obviously, a cloud-based PBX makes this easier since you get a dedicated IP address, and this Nerd Vittles article provides several options.

Once you receive your credentials, simply login to the TelecomsXchange web site. Just a few words about how the site is organized. Dashboard is where you’ll land when you login. Accounts let you specify more than one account to be associated with your credentials. If you manage multiple PBXs, this is where you set things up. Each account must have a unique IP address. This is also where you can change the IP address associated with your primary account if the need ever arises. My Interconnections displays each of your accounts and all of the carriers you’ve chosen to associate with each account. Market View is where you search for prices and choose carriers to associate with your account(s). We’ll cover this one in more detail a little later. Payment History tracks all of your payments to TelecomsXchange by date. Call Statistics lets you download CDR and Stats data by the day, week, or month. CDR gives you an instant snapshot of your calling history and the price of the calls based upon criteria you specify. It’s very similar to the same feature in the Incredible PBX or FreePBX GUI. Preferences let you change settings for your account.

The item you’ll need to use first is the plus sign (+) at the top of the form. This is how you fund your account. As noted previously, there is a 13% wholesale surcharge and this will be deducted from whatever amount you choose to add to your account. For example, if you add $100 using PayPal, the PayPal fee plus 13% will be deducted from the $100. So your account would show an available balance of approximately $84. Cash or bank wires also are accepted.

Finally, here’s a link to the TelecomsXchange Knowledge Base and Help Center. There’s lot of helpful information there to get you started.

Choosing a Carrier with TelecomsXchange

Let’s walk through the procedure to add new carriers to your account. This is the first thing you’ll want to do after you get your credentials and fund your account. Begin by making yourself a list of the countries or dialing prefixes you’d like to call using TCXC wholesale carriers. The easiest way to perform searches and find carriers is to decipher the dialing prefix for the calls you wish to make. For example, to call London, the Prefix would be the U.K. country code (44) plus the London city code (20).

Now open the Market View tab to get started. Here’s how we’d fill in the form to find London carriers and to order the first hundred matches from least costly to most expensive: Prefix (4420), Results (100), Order By (PRICE), and Route Type (CLI) which means you can specify your own CallerID for the outgoing calls. Click Search to proceed. The results look like this:

To add a carrier to your account, simply click on the plus sign (+) on the right side of the Action column beside the carrier of your choice. You then can choose whether to add it to all of your accounts, or you specify the account to which the carrier should be added. If you want to review the carrier’s history and ratings with TCXC, click on the Information icon in the Action column beside the carrier of your choice.

For NANPA call destinations, specify 1 plus the area code in the Prefix field. You can add the first 3 digits of the exchange to drill down further. Be advised that adding the 3-digit exchange may eliminate a number of carriers that only specify rates for an entire area code. For example, if a carrier specifies an area code rate for 1212 and no exchange limitations, then searching for 1212652 would not return that carrier.

If you already know which carriers you’d like to add, just search for them by specifying the carrier name in the Seller field and leaving the Prefix field blank. To get started, here are a few favorites for U.S./Int’l routes: IDT, LEXICO, TATA, VOXBEAM, and TELECOMSXCHANGE.

Placing Carrier-Specific Calls with TelecomsXchange

To set up the FreePBX Trunks and Outbound Routes, you first need to understand how calls are placed through TelecomsXchange carriers. In lieu of traditional trunk registrations on your PBX, TelecomsXchange uses the IP address that you registered for your account to determine whether SIP calls arriving at TCXC for routing to a carrier are authorized. Thus, it’s important that you keep your IP addresses updated whenever they change. Assuming your call passes the IP address check, the next hurdle is for TCXC to decipher which carrier should be used to route the call to its destination. This is handled by dialing prefixes which are unique to each TCXC carrier. For example, TATA has a dialing prefix of 32270#. To dial a U.S. call using the TATA carrier, the dial string would look like this: 32270#16785551212. A carrier must be assigned to your account before you can place calls from your PBX using that carrier’s dialing prefix. So there are two layers of protection on the TCXC side to prevent fraudulent calls. There must be both an IP address match and a carrier prefix match on your account before a call will be forwarded to a carrier.

Before we begin setting up your Trunks and Outbound Routes for Incredible PBX or one of the other Asterisk platforms, write down the names of each of the carriers you have chosen as well as their Dialing Prefixes. You’ll need them in the next steps. You can decipher carrier’s dialing prefixes assigned to your account under the My Interconnections tab in your TCXC Dashboard.

Setting Up TCXC Carrier Trunks in FreePBX

To begin, make certain that chan_SIP is assigned to UDP 5060 on your PBX. Particularly for trunks, there were just too many issues with PJsip in some releases of Asterisk so steer clear. With every TCXC carrier, the good news is the chan_SIP Trunk setup is virtually identical except for the carrier name and the carrier’s dialing prefix. For each carrier, start by adding a new chan_SIP Trunk in the Incredible PBX or FreePBX GUI. In the General tab, insert the carrier name in the Trunk Name field, e.g. TCXC. Leave the other default settings as they are.

Switch to the Dialed Number Manipulation Rules tab. Leave the Dialing Rules empty and insert the carrier’s dialing prefix in the Outbound Dial Prefix field, e.g. 77379#.

Switch to the SIP Settings tab. In the Outgoing tab, insert the carrier name in the Trunk Name field. Insert the following in the PEER Details field:

type=peer
qualify=yes
progressinband=never
port=5060
nat=yes
insecure=port,invite
ignoresdpversion=yes
host=sip01.telecomsxchange.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
allow=ulaw&alaw

 
While still in SIP Settings, switch to the Incoming tab, and clear out the default User Detail entries. Now click the Submit button and reload your dialplan when prompted.

Repeat this procedure for each of the carriers you set up in your TCXC profile.

Setting Up Outbound Routes for TCXC Calls

Our preferred Outbound Route setup for TCXC carriers is to create a new Outbound Route for each destination (typically a country) to which you wish to enable calling. Be advised that setting up a dialing prefix of just the number 1 authorizes considerably more calls than just those to destinations in the United States. For foreign countries, if all of your calls to the U.K are to destinations in London, then don’t authorize country-wide calling. Narrow it down to the country code and city code for London. Remember, it’s your phone bill.

For international calls, we prefer that callers enter a dialing prefix that specifies that it’s a long distance call plus a two-digit prefix representing the country abbreviation, not the dial code of the country. For example, for the U.K., we use 085 where 0 specifies long distance call and 85 is the phone representation for UK. We’ll then use the Outbound Route to strip off the caller’s dialed prefix and to insert the proper country code to complete the call.

Within each Outbound Route, we recommend you specify one or more low-cost carriers and a final TCXC carrier to catch calls that fail to all of your designated carriers. Otherwise, the caller will get a failed call. And you will get the next call. 🙂

So here’s what our Outbound Route setup for London, England looks like. You only need entries in the Route Settings and Dial Patterns tabs. Be sure to put your desired CallerID in the Route CID field and set the Override Extension option to YES. Then add your preferred Trunks in the order in which you want the calls attempted:

In the Dial Patterns tab, we specify a Prefix of 4420 to tell Asterisk to add a dialing prefix to the call to get it to London. Then we enter 085 in the Prefix field to tell Asterisk to strip off those digits entered by the caller before sending the call to the designated Trunk for processing. The Match Pattern is 8 X’s which represents an 8-digit London telephone number. To get a match on this Outbound Route, Asterisk will be searching for a dial string that looks like this: 085 + XXXXXXXX

Here’s an example of the Asterisk call flow using IDT as the primary trunk with this Outbound Route.

Caller Dials: 085-7499-0888
Outbound Route finds match on 085 Prefix + 8 X's and discards Prefix
Outbound Route sends 4420 (for London) + 74990888 to Trunk #1 (IDT)
IDT Trunk adds IDT Dialing Prefix 10729# before sending call to TCXC
TCXC receives: 10729#442074990888
TCXC strips IDT dialing prefix and sends call to IDT: 4420 + 74990888
IDT connects caller to Four Seasons Hotel in London

Adjustment with NAT-Based Implementations

Keep in mind that TCXC was designed primarily for commercial resellers, not for PBX-level implementations. If your PBX is sitting in the cloud or is directly connected to the Internet rather than sitting behind a NAT-based router, then you’re good to go now. If, on the other hand, your PBX is sitting on a private LAN behind a NAT-based router, make certain that your router forwards all UDP 5060 traffic to the private LAN address of your PBX. Otherwise, you may experience disconnect anomalies where the called party hangs up a call before your callers since there will be no call path for TCXC to return the disconnect alert (BYE) when the call is completed. For Incredible PBX servers, this isn’t really a problem because Incredible PBX will disconnect the call automatically after detecting 30 seconds of RTP traffic inactivity anyway. But we wanted to make you aware of the potential issue. The good news is you won’t be billed for the extra connection time since TCXC already has dropped the call with the carrier and turned off the billing meter.

Adding Trunk Information to Incredible PBX

Some may wish to include Trunk information in the CDR listings of Incredible PBX or FreePBX. This makes it much easier to spot problems when calls aren’t routed to the Trunk destinations you expect. It also makes it easy to generate trunk-specific reports within the GUI. In the FreePBX 12 and 13 implementations, the trunk information can be added painlessly by revising the [macro-dialout-trunk] context. However, you cannot make these changes directly in /etc/asterisk/extensions_additional.conf because your modifications will be overwritten the next time your dialplan is reloaded. Instead, the modified context must be added to extensions_override_freepbx.conf. Here’s how:

cd /tmp
wget http://incrediblepbx.com/cdr-trunk-info.tar.gz
tar zxvf cdr-trunk-info.tar.gz
rm -f cdr-trunk-info.tar.gz
cat cdr-trunk-info.txt >> /etc/asterisk/extensions_override_freepbx.conf
asterisk -rx "dialplan reload"

 
The modified CDR listing will look something like this:

We also developed a handy utility to make it easy to list out all of your trunks and their status. Here’s how:

cd /root
wget http://incrediblepbx.com/list-trunks.tar.gz
tar zxvf list-trunks.tar.gz
rm -f list-trunks.tar.gz
./list-trunks

 
The listing will look something like this:

Rate Queries Using the TCXC API

For those that want to query the TCXC rate tables locally, we’ve modified a TCXC sample JSON script slightly so that you can use Chrome (with JSONView) or FireFox (with JSON Lite viewer) to view JSON results. Using one of these browsers with the specified add-on, JSON results will be formatted automatically. The query results identity current providers and rates by entering a dialing prefix. The syntax for the web queries looks like the following where 192.168.0.224 is your server’s IP address and 357 is the dialing prefix rate table desired:


http://192.168.0.224/rates.php?prefix=357

 
The first 30 matching results will look something like this:


To use this script, you’ll need to insert your account name and API key (found in your TCXC Profile) into rates.php before first use. To install the script in the root folder of Apache, issue the following commands:


cd /var/www/html
wget http://incrediblepbx.com/TCXC-rates.tar.gz
tar zxvf TCXC-rates.tar.gz
rm -f TCXC-rates.tar.gz

 

Published: Monday, February 12, 2018



NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID. Enjoy!



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. We obviously have not verified TCXC’s billed minutes counter. Don’t rely upon it in deciding whether to use the service. As with all VoIP providers, what matters is the quality and cost of the calls. []
  2. Our special thanks to TelecomsXchange. They have generously offered to contribute a portion of the wholesale surcharge to support the Incredible PBX open source project. []

Rolling Your Own: Building a Custom ISO with Incredible PBX

We walked through the RedHat ISO creation procedure a couple years ago, but we wanted to revisit the issue using the 64-bit Scientific Linux 6.9 platform for those that want a current ISO image with Asterisk® 13 LTS. We recommend setting up your build environment on a VirtualBox virtual machine. Then you’ll have something to store away on your desktop computer in the event you ever want to create more customized ISOs. For today, you’ll need the 64-bit SL 6.9 DVD 1 ISO and the Nerd Vittles build environment and scripts. Should you ever decide to create all of the pieces from scratch, we’d recommend you begin with a careful review of Jason Priebe’s tutorial on Smorgasbork. But we’ll save you several hours of pain by providing the build environment and tools used to create the new Incredible PBX 13-13 ISO.

Getting Started. Begin by installing VirtualBox on your desktop computer. Then download the 64-bit SL 6.9 DVD 1 ISO. We chose Scientific Linux instead of CentOS or another RedHat derivative to minimize issues with copyrights and trademarks. Scientific Linux was partially developed with U.S. government employees and funding which imposes clear boundaries on attempts to limit redistribution of open source, GPL packages and packaging. Since we are merely adjusting the collection of GPL components in the ISO build and then adding our own GPL-licensed installation scripts in the kickstart files, the ISO is as close to worry-free as one can get in the litigious world in which we find ourselves.

Once you have created a virtual machine for Scientific Linux 6.9 and specified SL 6.9 DVD 1 ISO as your Virtual Drive in the Storage:Empty tree, start your VM and walk through the basic SL 6.9 install specifying a Minimal Install as your platform of choice. When the install finishes, reboot your server and login as root using SSH or Putty. Issue the following commands to add a few missing packages which we will need down the road:

cd /root
yum -y install net-tools nano wget tar
wget http://download.fedoraproject.org/pub/epel/6/x86_64/epel-release-6-8.noarch.rpm
rpm -Uvh epel-release-6-8.noarch.rpm

While it’s not normally good practice to build ISO images as the root user, we’ll do it anyway since we’ve built a virtual machine that will be dedicated exclusively to building ISO images. In the /root folder, the directory structure for our build environment will look like this:

/root
 -kickstart_build
  --isolinux
  --Packages  
  --repodata
  --images
    ---pxeboot 

Typically, most of the kickstart_build environment is populated from the isolinux directory on SL 6.9 disc 1. Then kickstart customizations are added to isolinux and the RPM packages to be loaded as part of the initial install are added to Packages. Finally, each of the kickstart files includes an opening install script, followed by a %packages section listing all of the main RPMs to be loaded (not the dependencies), followed by a %post section which specifies what tasks to complete once all of the RPM packages have been installed.

Our goodie bag for today sets all of this up for you. In addition, it adds a script (add-packages-tmp) which downloads all of the packages for the ISO as well as any required package dependencies. There’s also a text file (add-packages-kickstart) that lists all of the main packages that are going into the ISO build. This list will be used to populate the %packages section of each of the kickstart files (ks*.cfg) in isolinux. For the Incredible PBX 13-13 ISO, we’ve already moved the list of RPMs into each of the kickstart files so all you have to do is run /root/add-packages-tmp to download the RPMs and move them into place. Finally, there’s a script (create-ISO-new) in the /root folder that will actually generate the Incredible PBX 13-13 ISO file. Once it completes its tasks, the ISO file can be found in /root/kickstart_build.

Now let’s put the build environment and the Incredible PBX 13-13 ISO components into place and create our first ISO. While logged into your server as root, issue the following commands. NOTE: Over 1,000 packages make up the Incredible PBX 13-13 ISO so the add-packages-tmp script takes a good long while to run since each RPM has to be downloaded and moved into place in our build environment.

cd /root
wget http://incrediblepbx.com/create-ISO-6.9.tar.gz
tar zxvf create-ISO-6.9.tar.gz
./add-packages-tmp
./create-ISO-new
ls /root/kickstart_build/*.iso

To add your own RPMs to an ISO, simply edit add-packages-tmp and add each new RPM in two places, in the reinstall AND the install sections of the script. This assures that RPMs will be added whether they already exist on your virtual machine or not. For each new RPM you add, be sure to also add an entry in add-packages-kickstart. We prefer to keep the packages alphabetical which makes it easier to make changes down the road. Finally, cut and paste the add-packages-kickstart list into the %packages section of each of your kickstart files in kickstart_build/isolinux/ks*.cfg. The isolinux.cfg file is where you set up the install menu for the ISO and specify the names of the kickstart files to associate with each menu selection. Once you’ve customized things, simply rerun add-packages-tmp and create-ISO-new to generate a new ISO. Enjoy!

Published: Monday, January 29, 2018



NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 10-digit DID. Enjoy!



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…