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The Most Versatile VoIP Provider: FREE PORTING

Now Serving: The Incredible PBX 13-13 Whole Enchilada

We’re delighted to introduce the bells and whistles for Incredible PBX® 13-13. We’ve taken a slightly different approach with this release. Instead of getting the Whole Enchilada out of the box, you now have a choice. You start with Incredible PBX 13-13 LEAN on the recommended CentOS® 6.10 platform. This gets you a fully-functioning PBX with the latest Asterisk® 13 and most of the FreePBX® 13 GPL modules. This release includes support for Skyetel SIP trunking from our Platinum Sponsor together with $50 in free service to get you started. You still can customize your PBX in any way you like. Or just upgrade to the Whole Enchilada and take advantage of the entire feature set that Incredible PBX has traditionally offered. Last but not least, you can add Incredible Fax for flawless faxing with HylaFax® and AvantFax® including fax detection on specified inbound routes. So the choice is totally up to you. We have a lot to cover. For today, we’ll get all the Incredible PBX pieces installed.

Just Released: Incredible PBX 16-15 for CentOS 7. Take it for a test drive.

Here’s a sneak peek at what’s included in Incredible PBX 13-13 Whole Enchilada: dozens of preconfigured SIP Trunks from our favorite providers, Voice Dialing (411) with IBM STT or Google, Headline News (951), Weather by ZIP Code (947), Today in History (86329), IBM TTS, ODBC Lookups (222), ODBC Calling with AsteriDex (223), Telephone Reminders (123), AsteriDex (Web GUI), Reminders (Web GUI), PortKnocker, Travelin’ Man 4, Time of Day (*61), SMS Dictator (767), Wolfram Alpha (4747), Hotel-Style Wakeup Calls (*68), Allison’s Demo IVR (3366), Lenny (53669), Call Parking (**70), Call Pickup (71), Blacklist Add (*30), Blacklist Remove (*31), Blacklist Last Caller (*32), Call Forward Activate (*72), Call Forward DeActivate (*73), Conferencing (C-O-N-F), Call Pickup (*8), Dictation (*34), Email Dictation (*35), DND Activate (*78), DND DeActivate (*79), SpeedDial with AsteriDex (000NNN), Email Delivery of Voicemails, NeoRouter VPN, and more. With a little luck, this will light a fire under some of you to roll up your sleeves and participate in the open source development community.

Installing a Base CentOS Operating System

You can install Incredible PBX 13-13 Lean on a dedicated server, on a virtual machine platform such as VirtualBox, or a Cloud-based server. We recommend a minimum 1GB of RAM with a swapfile unless installing on OpenVZ platforms. We’ve provided a script to do it for you. Depending upon the number of users your server will be supporting, we recommend a disk capacity of 10-30 GB. Last but not least, you need a reliable Internet connection.

Before you can install Incredible PBX 13-13 Lean, you’ll need a basic Linux platform. For this build, you can start by deploying a minimal install of CentOS 6. The Incredible PBX installer will load all of the necessary components to support Asterisk and FreePBX as well as upgrading CentOS to 6.10. Better yet, use the new Incredible PBX 13-13 ISO which bundles both the operating system packages and all of the Incredible PBX goodies. Complete Incredible PBX 13-13 ISO tutorial available here.

Begin by installing 64-bit CentOS 6 on your favorite hardware or Desktop. Or you may prefer to use a Cloud provider1 that already offers a preconfigured CentOS or Incredible PBX 13-13 image in the case of HiFormance. If you’re using a Cloud platform, you can skip the rest of this section. Just choose CentOS 6 or Incredible PBX 13-13 on HiFormance as the default operating system for your cloud-based server.

For those using a dedicated hardware platform or wishing to install CentOS as a virtual machine, the drill is the same. Start by downloading the 64-bit CentOS 6.10 minimal ISO. Burn the ISO to a DVD unless you’ll be booting from the ISO on a virtual machine platform such as VirtualBox. On virtual platforms, we recommend at least 1GB RAM and a 20GB dedicated drive. For VirtualBox, we’ve provided a one-minute installer. Here are the settings:

Type: Linux
Version: RedHat 64-bit
RAM: 1024MB
Default Drive Options with 20GB space
Create
Settings->System: Enable IO APIC and Disable HW Clock (leave rest alone)
Settings->Audio: Enable
Settings->Network: Enable, Bridged
Settings->Storage: Far right CD icon (choose your ISO)
Start

If you’re booting your server with the CentOS ISO to start the CentOS install, here are the simplest installation steps:

Choose Language and Click Continue
Click: Install Destination (do not change anything!)
Click: Done
Click: Network & Hostname
Click: ON
Click: Done
Click: Begin Installation
Click: Root Password: password, password, Click Done twice
Wait for Minimal Software Install and Setup to finish
Click: Reboot

Installing Incredible PBX 13-13 LEAN

Unless you’re using a virtual machine Incredible PBX image or the Incredible PBX 13-13 image on HiFormance, you’ll need to run the Incredible PBX installer. Once you have CentOS up and running, log into your server as root and issue the following commands to kick off the Incredible PBX install.

passwd
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
# to add swap file on non-OpenVZ cloud platforms
./create-swapfile-DO
# kick off Phase I install
./IncrediblePBX-13-13.sh
# after reboot, kick off Phase II install
./IncrediblePBX-13-13.sh
# add Full Enchilada apps, if desired
./Enchilada-upgrade.sh
# add HylaFax/AvantFax, if desired
./incrediblefax13.sh
# set passwords
./update-passwords
# set desired timezone
./timezone-setup
# remember to enable TUN/TAP if using VPS Control Panel
# reconfigure PortKnocker if installing on an OpenVZ platform
echo 'OPTIONS="-i venet0:0"' >> /etc/sysconfig/knockd
service knockd restart
# fix pbxstatus for NeoRouter VPN support, if desired
cd /usr/local/sbin
sed -i "s|cat /etc/hostip|cat /etc/hostip \\| cut -f 3 -d ' ' |" pbxstatus
# set up NeoRouter client, if desired
nrclientcmd

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

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

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.

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

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. If you have an earlier release, you may need to download the Whole Enchilada tarball as documented below. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Try issuing just the last command first to see if the enchilada upgrade script already is in place. Otherwise, execute all of the commands below. Be advised that the 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.

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:

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

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. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.

Getting Started with IBM Watson TTS Service

We’ve created a separate tutorial to walk you through obtaining and configuring your IBM Watson credentials. Start there.

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. For new users, your $IBM_username will be apikey. Your $IBM_password will be the TTS APIkey you obtained from IBM. Next, 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.

Getting Started with IBM Watson STT Service

Now let’s get IBM’s Speech to Text service activated. Log back in to the IBM Cloud. Click on the Speech to Text app. Choose a Region to deploy in, choose your Organization from the pull-down menu, and select STT as your Space. Choose the Standard Pricing Plan. Then click Create. When Speech to Text Portal opens, click the Service Credentials tab. In the Actions column, click View Credentials and copy down your STT username and password.

Finally, login to your Incredible PBX server and issue these commands to edit getnumber.sh:

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

Insert apikey as your API_USERNAME and your actual STT APIkey 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.

Adding Skyetel Trunks to Incredible PBX

Now that you have your Incredible PBX platform in place, it’s time to set up your Skyetel trunks to take advantage of the BOGO calling credit (up to $250). The trunks themselves are added by logging into your server with SSH/Putty as root and issuing the following commands if the trunks aren’t already installed on your server. HINT: Check first!

cd /root
wget http://incrediblepbx.com/add-skyetel
chmod +x add-skyetel
# uncomment next line if your incoming calls all have 10-digit numbers
# sed -i 's|from-trunk|from-pstn-e164-us|' add-skyetel
./add-skyetel
chmod -x add-skyetel

Next, sign up for Skyetel service and take advantage of the exclusive Nerd Vittles BOGO offer. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Skyetel will match your original deposit up to $250 which means you could enjoy as much as $500 of SIP trunking service for half price. Effective 10/1/2023, $25/month minimum spend required. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request your BOGO credit by referencing this Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print.2 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!

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

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: server1.incrediblepbx.com

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

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

Configuring a Skyetel Inbound Route

Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.

If you have installed the Incredible Fax add-on, you can enable Fax Detection under the Fax tab. And, if you’d like CallerID Name lookups using CallerID Superfecta, you can enable it under the Other tab before saving your setup and reloading your dialplan.

Configuring a Skyetel Outbound Route

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose Connectivity -> Outbound Routes -> Add Outbound Route. For the setup, we recommend the following using the CallerID Number you wish to associate with your outbound calls through Skyetel:

Enter the Dial Patterns under the Dial Patterns tab before saving your outbound route. Here’s what you would enter for 10-digit and 11-digit dialing. If you want to require a dialing prefix to use the Skyetel Outbound Route, enter it in the Prefix field for both dial strings.

There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes for the remaining trunks.

Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.

HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.

Audio Issues with Skyetel

If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:

externip=xxx.xxx.xxx.xxx

If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.

Receiving SMS Messages Through Skyetel

Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS & MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:



Sending SMS Messages Through Skyetel

We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create an SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.

Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.

cd /root
wget http://incrediblepbx.com/sms-skyetel
chmod +x sms-skyetel
nano -w sms-skyetel

To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"

Using Gmail as a SmartHost for SendMail

Many Internet service providers block email transmissions from downstream servers (that’s you) to reduce spam. The simple solution is to use your Gmail account as a smarthost for SendMail. Here’s how. Log into your server as root and issue the following commands:

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

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

Now issue the following commands:

chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
service sendmail restart

Finally, send yourself a test message. Be sure to check your spam folder!

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

Check mail success with: tail /var/log/maillog. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.

Originally published: Monday, November 13, 2017  Updated: Saturday, March 23, 2019


News Flash: Turn Incredible PBX into a Fault-Tolerant HA Platform for $1/Month

Continue Reading: Configuring Extensions, Trunks & Routes

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

Check out the new Incredible PBX 13-13 ISO. Complete tutorial available here.


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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. Some of our links refer users to Amazon or other service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. However, when pricing is comparable or availability is favorable, we support these providers because they support us. []
  2. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Keep On Trunkin’: Free International VoIP Calling Returns



Today we’re taking a fresh look at the international calling marketplace by updating the best VoIP deals available. FreeVoipDeal once again takes the prize with the best selection of "free" international calling destinations at the lowest prices. Below we’ll provide a quick tutorial to transform your Incredible PBX server into an international calling platform at minimal cost.

Here’s How It Works. For every 10 euros ($10.72) you deposit into your account, you’ll get 300 minutes a week of free calls to a specific list of countries for 120 days. After you exhaust your free minutes, calls to the "free" countries revert to their standard VoIP rates. You can also call anywhere else in the world at very reasonable per minute rates that compare favorably with other SIP providers around the world. The beauty of a PBX and SIP trunks is you can mix and match as many providers as you like to take advantage of favorable calling rates to multiple countries. We’ll walk you through the FreeVoipDeal trunk setup below.

Betamax 101. There are a few things you need to know about the so-called Betamax VoIP services up front. Most importantly, they change rates and free countries more frequently than college kids change partners. The calling rate to some country from some Betamax provider changes almost every day because Betamax has dozens of companies offering similar services with differing rates and freebies. Here’s an very old spreadsheet that will give you a good idea of what you’re up against. Don’t depend upon it for the current rates. You’ll need to visit the actual site(s) for their current rate tables or visit this site (not) maintained by Betamax for a country-by-country comparison by provider. That’s another way of saying DON’T BLAME US IF YOUR 3-HOUR CALL TO ANTARCTICA CHANGED FROM 20¢ PER MINUTE TO $1 PER MINUTE OVERNIGHT. IT PROBABLY WON’T, BUT IT MIGHT.

One other word of warning. Some Betamax sites (marked with a red asterisk in the Betamax country table) such as powervoip.com have good calling rates, but they tack on a 3.9¢ connection fee to every call. If you make lengthy calls, it’s not a big deal. If you make numerous short calls, it drives your discount calling rates through the roof. Before making a lengthy call to a remote destination, spend the two minutes it takes to look up the current rate on the actual Betamax web site and take a snapshot of the page for your records. Here’s another tip. If you make frequent calls to Antarctica, spend a little time doing your homework. Review the latest Betamax spreadsheet to track down the cheapest rates. Then double-check the actual sites for the current rates. There’s a $100+ difference in the cost of a 3-hour call at €.20/minute from some Betamax sites versus the €.70/minute rate at some other Betamax sites. THIS OFTEN CHANGES! HINT: Don’t use FreeVoipDeal for Antarctica.

Today we’ll be focusing on the company we’ve tracked for many years, FreeVoipDeal.com. Except for the domain name, the setup with other Betamax providers is similar but not identical. And, of course, you’ll have to kick in another deposit to make free calls from each site. The length of the Freebie period also may vary so read the terms carefully. FreeVoipDeal actually hasn’t changed much since our first visit about five years ago. In fact, we still had most of our ten euro credit so we could play all we wanted even though the calls were no longer free since our four month window has long since expired.

Here’s the February 23, 2019 Freebie list by country. Don’t depend upon it! Check their actual web site or the Betamax country summary for current freebies and current rates. Here’s a great trick to remember. When you visit the FreeVoipDeal Rate Table, click on the Out of Minutes tab for a quick listing of all the Free Calling Countries as well as the rates once you’ve used up your four months or 300 weekly minutes of free calls. With few exceptions, most of the "free countries" still have a rate of 1.1¢ per minute even after you run out of minutes.

 

How Free International Calling Works

Placing international calls through FreeVoipDeal can be done in a number of ways. That’s the real beauty of a PBX. First, you can either load an app to make the calls if your smartphone or PC supports it. With Incredible PBX, you can use a SIP phone to dial a FreeVoipDeal number directly through your PBX, or you can dial a DISA access number or SIP URI from anywhere to connect to your PBX and then enter your DISA password after which you will get a second dial tone to place an international call using your FreeVoipDeal trunk. The beauty of the DISA approach is you can call into your PBX from any telephone to place free or dirt cheap international calls.

Using Incredible PBX 13 and DISA for Calling

On the Incredible PBX platform, you can use the DISA application to provide secondary dialtone for processing international calls. A phone number and trunk will receive incoming calls bound for DISA from your cellphone. An inbound route will only forward incoming calls to DISA that match your cellphone number. A secondary trunk from FreeVoipDeal or other providers will be used to process outgoing international calls that are dialed using DISA. We’ll create an outbound route or rule for every country to which you want to authorize international calling. Each of these outbound routes will point to the least expensive (or free) trunk to complete the call. In the VoIP world, you actually could have dozens of outbound trunks that handle international calls based upon the country codes of each international call. This lets you take advantage of the best calling rates for each country. We will block international calls to country codes not specifically authorized.

Just to restate the obvious, a misconfigured DISA application that allows the world to make international calls on your nickel can get expensive quickly. We’ll protect today’s DISA setup for Incredible PBX with three layers of protection. First, we’ll require that the CallerID of the incoming call match your cellphone number. While this isn’t failsafe since CallerID numbers can be spoofed, it does reduce the risk considerably. Second, to make DISA calls, you’ll have to know the incoming phone number or SIP URI managing DISA on your PBX. And third, you’ll have to enter the correct DISA PIN before being prompted for an international number to dial. Without all three, nobody gets to make an international call on your nickel. Just remember, compromising DISA on your PBX is just as risky as handing out your credit card to a stranger so follow the setup steps below carefully. And then TEST, TEST, TEST to make sure strangers can’t access your DISA setup. We’ll show you how.

Here’s an overview of the DISA setup drill once you have Incredible PBX running. We’ll walk through each of the six steps below. Don’t get frustrated. There are a number of steps, but none of them are difficult. Just pretend you’re baking cookies and don’t skip any steps.

  1. Set Up Your Trunk to Process Incoming DISA Calls
  2. Set Up Your Trunk(s) to Process Outgoing International Calls
  3. Configure DISA with a Very Secure Password
  4. Configure an Inbound Route to Limit Incoming DISA Calls to Your Cellphone #
  5. Configure an Outbound Route for Each International Country Code
  6. Test, Test, Test

1. Setting Up Incoming DISA Call Trunk

Before you can make calls to your PBX, it’ll need a phone number (known affectionately as a DID). As installed, Incredible PBX includes preconfigured SIP trunks from about a dozen SIP providers. All you’ll need is credentials from the company you wish to use. You can obtain a free DID here. To obtain your own SIP URI, read our tutorial.

2. Trunk Setup for International Calling

We’re going to walk you through setting up a trunk with FreeVoipDeal to handle free international calls to certain countries documented above. This may not be the best fit for you depending upon the international destinations you wish to call. Figure that out first! Then adjust the trunk settings below to match each SIP provider trunk you wish to create. There’s no limit to the number you can have. And, with most of these providers, you pay by the minute for international calls anyway so there is no harm in configuring multiple trunks to take advantage of the best rates calling the countries of your choice. The same applies to all-you-can-eat and "free" trunks except there are varying fees for using the services so you’re probably not going to want a dozen of them even if some of the calls are free after making a periodic deposit. Start with the pink and green entries on the old spreadsheet we referenced for the cheapest historical rates and then visit the actual sites and read the fine print.

To add new trunks to Incredible PBX, use a browser to access the IP address of your server. Login with the default username of admin and the password that you set when your install completed. You can change it with the admin-pw-change script in /root. Once the dashboard appears, click the Connectivity tab and choose Trunks -> Add SIP (chan_sip) Trunk.

For Trunk Name, enter FreeVoipDeal. In the Dialed Number Manipulation Rules section, add a rule for each country code you wish to activate. You can decipher the Country Code for any country at this link. For example, for the United Kingdom, you’d enter a rule like this where 44 is the Country Code and each X represents a required digit in the local area code and phone number. The trailing period means the number includes one or more additional digits. NOTE: DISA calls will not have to be prefixed with 011 to place international calls. Just enter the country code and number to be called. And, we are told that only 441, 442, and perhaps 443 calls to the U.K. are free since those are the designated landline prefixes.




If there are other countries, you wish to support with this trunk provider, you’d click Add More Dial Pattern Fields and insert an additional rule for each country following the example above. If you’ll be using this trunk to make calls in the U.S. and Canada as well, the correct Match Pattern is 1NXXNXXXXXX, and calls will need to be dialed with the 1 to avoid conflicts with international dialing.

Next, we need to enter the Outgoing Settings. For the Trunk Name, enter freevoipdeal. Clear out the entries in Peer Details section and enter the following using your actual FreeVoipDeal credentials for yourusername and yourpassword:

authuser=yourusername
username=yourusername
secret=yourpassword
type=peer
qualify=yes
nat=yes
insecure=port,invite
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
dtmfmode=auto
disallow=all
canreinvite=no
allow=alaw&ulaw

Finally, clear out the default entries in User Details and click the Submit Changes button and then red Apply Config button to save your new trunk.

Spoofing Your CallerID. When setting up your FreeVoipDeal account, you can set up one or more numbers to use as your CallerID number on FreeVoipDeal calls. You simply verify the number with a code sent by SMS or phone call from their service. Once you’ve gone through the verification procedure, you can spoof the outbound CallerID on FreeVoipDeal calls using your actual cellphone number. Just add the following entries to your Trunk settings replacing 9991234567 with your cellphone number. Special thanks to @hillclimber on the PIAF Forum for the tip.

fromuser=0019991234567
sendrpid=yes

3. Configuring DISA for International Calling

In the Incredible PBX GUI, we’ll set up DISA by clicking the Applications tab and choosing DISA. Add your new DISA configuration by following this sample. Use a VERY secure password. It’s your phone bill. Once you’ve finished, click the Submit Changes button and then the Apply Config button to save your new DISA setup.



4. Inbound Routing of DISA Calls

Here’s where we lock down your setup so that Incredible PBX only accepts DISA calls from your cellphone number. If you want to allow additional people to use your DISA setup or if you have multiple cellphones, then simply create multiple inbound routes with the 10-digit numbers of each phone to be supported.

In the Incredible PBX GUI, we’ll set up a new Inbound Route by clicking the Connectivity tab and choosing Inbound Routes. If you plan to support multiple phones, then create multiple inbound routes and give each of them a unique Description and CallerID Number that matches the phone number of the cellphone to be supported. Be sure to check the CID Priority Route checkbox and set the correct Destination for your incoming calls. Just fill in the blanks appropriately using this template as a guide. Once you’ve finished, click the Submit button and then the Apply Config button to save your new Inbound Route.



5. Outbound Routing by Country Code

The DISA application is going to obtain the phone number to be dialed and will pass that to the Outbound Routes module. The job of the Outbound Routes module is to examine the phone number passed to it from DISA to figure out which trunk to use to make the outbound call. It then will pass the call to the appropriate trunk which sends the outgoing call on its way to the destination.

For each Dialed Number Manipulation Rule in every Trunk that you set up in Step #2 above, you’ll need a matching Outbound Route if your PBX is used to place calls using multiple trunks. If you’re only using one provider for all of your outbound calls, then we can use a more generic Outbound Route. It’s always a good idea to create the one-to-one match between Outbound Routes and Trunks to make certain that outbound calls are sent to the correct Trunk for processing. So let’s do that using the U.K. trunk we created above.

In the Incredible PBX GUI, we’ll set up a new Outbound Route by clicking the Connectivity tab and choosing Outbound Routes. When the template appears, notice in the far right column that there’s a listing of all your existing Outbound Routes. Calls are actually processed sequentially using the order that these Outbound Routes appear in the list. If there’s no number match in the top route, processing drops to the next route in the list until there is a match AND a successful connection. You can adjust the sequence by dragging the Outbound Routes to a different position in the priority list.

It’s important to use specificity in your Outbound Routes (especially with International calling) to make certain that a call isn’t inadvertently processed by some other trunk. The easiest way to do this is to require the Outbound Route Match Pattern for U.K. calls to be at least 11 digits, e.g. 44XXXXXXXX. (the trailing period is important in that it requires at least one more digit for a match). And we can force a Hangup if the FreeVoipDeal trunk is not available for some reason by adjusting the Destination on Congestion setting. This keeps the call routing from dropping down to the next available Outbound Route in the list if FreeVoipDeal happens to be off-line at some point. So our Outbound Route for U.K. calls should look something like this:



The final step is to move the new Outbound Route for U.K. calls to the top of the Outbound Routes listing in the right column to assure that it is processed first. Once you’ve done that, click the Submit Changes button and then the Apply Config button to save your new Outbound Route AND the adjusted Outbound Route Priority List.

Another alternative in creating Outbound Routes is to use a Dial Prefix that never matches a real phone number to direct calls to a particular trunk. For example, you might use *8 as a dial prefix for FreeVoipDeal calls. By placing *8 in the Prefix column of the Dial Pattern, it will get stripped off before the number is actually passed to the FreeVoipDeal trunk for processing. We actually prefer this setup because it adds an additional layer of security for international calls. If someone were to break into your DISA application by knowing your cellphone number AND your DID AND your DISA password, it’s unlikely they’d also know to prefix outgoing international calls with some arbitrary dial prefix. Just don’t use *8 in case they’re a Nerd Vittles reader. 😉

6. Test, Test, Test!

The easiest way to test the new setup is to place a couple of calls and to watch the Asterisk CLI (asterisk -rvvvvvvvvvv) and see how the calls are processed and who answers at the other end. Then you can apologize for reaching the wrong number.

You can make up your own test methodology, but here’s one that works for us. There are several tests you need to make. First, call your Incredible PBX DID from your authorized cellphone and enter a correct DISA password to see if you get dial tone to make an international call. Then repeat the drill with an invalid password and make sure you don’t get a dial tone. Next, call your Incredible PBX DID from a phone other than your authorized cellphone. You should not get a prompt for a DISA password. Finally, we use the first three digits of a U.K. number to identify a matching NANPA area code. Then, we find hotels in the two matching cities. For example, one might attempt to call a hotel in Bath, England (44 1… ……) and a hotel in Bermuda (441-…-….). The U.K. call should go through, and the Bermuda call should fail. If you pass all three tests with flying colors, you’re good to go.

Using FreeVoipDeal’s MobileVoIP App

FreeVoipDeal also offers a MobileVoIP app that can be used directly on your smartphone (Android, iOS, and Windows phone versions available) using any Wi-Fi, UMTS, 4G/LTE, 3G, GPRS or EDGE connection. The drawback is the lack of the three extra layers of security protection that Incredible PBX using DISA offers. MobileVOIP lets you log in with your registered Betamax credentials and offers the option to use your existing VoIP credit from your smartphone. The downside is that anyone with the app and your credentials can call anywhere and talk for as long as they like on your nickel using any of your registered CallerIDs. You’ve been warned. For more information or to download the app for your mobile device, go here. Remember to dial the "+1″ country code prefix for U.S./Canada calls.

Originally published: Monday, April 24, 2017  Updated: Monday, February 25, 2019


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 VoIP-info Forum.


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Some Recent Nerd Vittles Articles of Interest…

Big Kahuna: 70 New FreePBX GPL Modules for Incredible PBX



We don’t change the mix of FreePBX® GPL modules in Incredible PBX® 13-13 often although you can easily add or update any of particular interest at any time using the gpl-install scripts included in the distribution. So we’re excited to introduce the 2019 collection of 70 FreePBX GPL modules for those that want to keep their Asterisk® PBX platform loaded with the latest and greatest. We’ve included a batch installer which means ALL of the existing modules get updated with the latest releases from GitHub. Depending upon the speed of your Internet connection, it’s a 5 or 10-minute procedure. Schedule it for a time when the PBX is idle.

Upgrading the FreePBX GPL Modules

The upgrade procedure couldn’t be easier. Log into your server as root. We recommend you make a backup first using the incrediblebackup script in /root. Next, make sure you have at least 50MB of free disk space: df -h. Then issue these commands and have a cup of coffee:

cd /tmp
wget http://incrediblepbx.com/modules13.tar.gz
tar zxvf modules13.tar.gz
rm -f modules13.tar.gz
cd modules13
./update-modules.sh




When you return, your Incredible PBX 13-13 server will be all shiny and new. You can review the license terms for each module by referencing the table below and calling up the GPL license provisions with a browser pointed to http://server-IP/admin/licenses.



FreePBX GPL Modules Documentation

The FreePBX Dev Team has generously provided excellent documentation for all of the modules. We have arranged them in the same order as the GUI’s menus for ease of use.

   

Admin Modules

Administrators Module
Asterisk CLI Module
Asterisk Phonebook Module
Backup and Restore Module
Blacklist Module
Bulk Handler
CID Superfecta
CallerID Lookup Sources
Certificate Management Module
Config Edit
Custom Destinations Module
Custom Extensions Module
Feature Codes Module
Module Admin Module
Phone Restart Module
Presence State Module
REST API
Sound Languages
System Recordings Module
User Management Module

Applications Modules

Announcements Module
Call Flow Control Module
Call Recording Module
Callback Module
Conferences Module
DISA Module
Directory Module
Extensions Module
Follow Me Module
IVR Module
Languages Module
Misc Applications Module
Misc Destinations Module
Paging and Intercom Module
Parking Module
Queue Priorities Module
Queues Module
Ring Groups Module
Set CallerID Module
Text to Speech Module
Time Conditions Module
Time Groups Module
Voicemail Blasting Module
Wakeup Calls Module

Connectivity Modules

DAHDI Channel DIDs
Inbound Routes Module
OSS End Point Manager (Disabled)
Outbound Routes Module
Trunks Module

Dashboard

PBX System Status

Reports Modules

Asterisk Info Module
Asterisk Logfiles
CDR Reports Module
Call Event Logging (CEL) Module
Print Extensions
Rest API Report
Weak Password Detection

Settings Modules

Advanced Settings
Asterisk IAX Settings
Asterisk Logfile Settings
Asterisk Manager Interface
Asterisk SIP Settings
Extension Settings
Fax Configuration
Music on Hold Module
Pin Sets
Route Congestion Messages
Text to Speech Engines Module
Voicemail Admin

Third Party Addons

AsteriDex
Reminders

UCP

User Control Panel

Installing OSS Endpoint Manager

If you have dozens of SIP phones to configure, then you’ll appreciate Andrew Nagy’s terrific OSS Endpoint Manager Module. Here’s how to install it once your Incredible PBX 13-13 server is updated with the new modules above:

cd /
wget http://incrediblepbx.com/epm.tar.gz
tar zxvf epm.tar.gz
./install-epm.sh

You will also need to install and configure a TFTP server. Here’s the CentOS procedure:

cd /root
wget http://incrediblepbx.com/setup-tftp
chmod +x setup-tftp
./setup-tftp

Pay particular attention to the firewall instructions which display at the end of the TFTP install procedure. Complete documentation for OSS Endpoint Manager is available here. Enjoy!

Originally published: Monday, February 18, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Introducing Skyetel: A VoIP Provider for All Seasons

Having been around the block more times than we can remember, suffice it to say it takes a lot to get us excited about a VoIP provider. Let us tick off some criteria to even get our attention: terrific pricing, failsafe reliability, and first class performance. So just imagine our excitement to discover that an early follower of Nerd Vittles now provides one of the most compelling VoIP services we’ve ever tested with triple redundancy in multiple data centers. And Skyetel now has added what, for some, was the most important piece: support for VoIP servers with dynamic IP addresses. While it’s still beta code, it’s easy to use and reliable. There’s yet another hidden benefit. Incredible PBX coupled with Skyetel makes a perfect platform for redundant servers. We’ll cover it in a future article, but here’s the basic design.

Let’s sweeten the pot a bit more. We were looking for a service provider that could offer a compelling price for the hobbyist and home user while also having the depth to provide millions of minutes to organizations and resellers that actually have such a need. Skyetel now offers Nerd Vittles readers two special offers. First, you can claim a $10 credit for your new account simply by opening a ticket once you sign up. Once you have kicked the tires and are satisfied with the service, you won’t want to miss the Nerd Vittles BOGO offer. Skyetel will match your original deposit up to $250. Deposit $50 and Skyetel will double it. Or plan ahead with a $250 deposit and Skyetel will still double it. That translates into $500 of half-price VoIP service! Once you have funded your account with your money, Skyetel will provide free porting of your DIDs for the first 60 days after you open your account plus a 10% reduction in your current origination rate and DID costs by presenting your last month’s bill.1 Effective 10/1/2023, $25/month minimum spend required. For resellers and high volume users, document your requirements on your Nerd Vittles signup form and let us put you in touch with someone at Skyetel that will make you a deal you can’t refuse. And what does Nerd Vittles get out of this? Glad you asked. We’re delighted to have Skyetel as a platinum sponsor to keep the lights burning and the deals flowing for another decade of articles and open source offerings for our dedicated followers.

Original Skyetel DepositSkyetel Deposit MatchAvailable SIP Service $'s
$20$20$40
$50$50$100
$100$100$200
$200$200$400
$250$250$500

We want to also address the elephant in the room. Some have asked about our relationship with Vitelity, a long time sponsor of Nerd Vittles and our open source projects. They’re alive and well. However, the company has gone through several acquisitions in the past few years, and their focus now has shifted more to the reseller and wholesale market. ALL EXISTING VITELITY CUSTOMERS ARE UNAFFECTED BY THIS CHANGE IN DIRECTION. And we are more than happy to put new resellers and wholesalers in touch with someone at Vitelity that can address your requirements. The good news is that you’ll now have two companies to compare while new home users and small businesses have a viable alternative moving forward.

Skyetel’s State-of-the-Art Network Design

Because Skyetel’s system architecture is radically different from most other VoIP providers, we wanted to spend a minute documenting their setup. Typically, a VoIP provider may offer a failover server in case their primary server fails. But all calls flow through the primary server unless there is a system failure. As we noted previously, Skyetel’s current setup includes three redundant data centers, all of which receive incoming calls while being firewalled from each other. Once you place or receive a call from the Skyetel network, their data center is completely removed from the audio path of the call which flows directly between your server and the outside party. Thus, even if the data center experienced a total system failure in the middle of your call, neither you nor the other party would ever know it. This design also eliminates the potential of a man-in-the-middle attack from your VoIP provider’s server.

Skyetel Pricing Overview

This summary is not intended to be an exhaustive listing of all Skyetel services. Follow this link for a complete summary of fees and services. Traditional DIDs are $1 per month. Toll free numbers an additional 20¢ per month. Outbound conversational calls are $0.012 per minute. DIDs can be SMS/MMS enabled for 10¢ per month. E911 service is $1.50 per month. Incoming conversational calls are a penny a minute. CallerID lookups are $0.004 per call. Voicemail transcription is available for 10¢ per message.

Signing Up for Skyetel Service

So here’s the drill to sign up for Skyetel service and take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request your free $10 credit to kick the tires. You cannot port in numbers at no cost until you actually fund your account out of your own pocket. Once you have funded your account, open another ticket for the BOGO credit for your account by referencing the Nerd Vittles special offer. You then can initiate your free number porting requests on the portal and request a credit for the porting fees. BOGO credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, attach a copy of your last month’s bill. See footnote 1 for the fine print. If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. Easy Peasy!

For those that may be concerned that one day, after your credit expires, you could be paying a penny a minute for phone calls, let me provide a little Ma Bell history lesson for you. When my roommate and I were in law school, our typical phone bill often exceeded $200 a month because we both had girlfriends a couple hundred miles up the road. In today’s dollars, that phone bill translates into roughly $1,200 a month. That would have been 120,000 minutes a month at a penny a minute in today’s dollars. So, yes, VoIP is having a profound influence on the AT&T and Verizon Bell Sisters.

Skyetel Endpoint Group Configuration

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

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: server1.incrediblepbx.com

Skyetel DID Configuration

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

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

Incredible PBX Firewall Setup for Skyetel

The Travelin’ Man 3 firewall included with all Incredible PBX platforms limits access to your server based upon whitelisted IP addresses of outside providers and users. In order to receive calls from the multiple Skyetel data centers, the following entries need to be included in the whitelist of your PBX. For new installs of Incredible PBX 13-13 for CentOS, the entries already are included. Otherwise, issue the following commands from the Linux CLI and choose the 0 option using the add-ip utility in /root:

  • /root/add-ip Skyetel-NW 52.41.52.34
  • /root/add-ip Skyetel-SW 52.8.201.128
  • /root/add-ip Skyetel-NE 52.60.138.31
  • /root/add-ip Skyetel-SE 50.17.48.216
  • /root/add-ip Skyetel-EU 35.156.192.164

NOTE: If your PBX is sitting behind a NAT-based router, then you will also need to forward UDP port 5060 from your router to the internal IP address of your PBX. Otherwise, incoming calls from Skyetel will fail. You also may need to add a NAT=yes entry to each of the Skyetel trunk configurations using the GUI. The telltale sign that the NAT entry is required will be incoming calls with one-way or no audio.

Incredible PBX Trunk Setups for Skyetel

Because Skyetel uses multiple data centers without trunk registrations, you’ll actually need to configure 6 separate Skyetel trunks in the Incredible PBX GUI. The same setup applies for those using generic FreePBX aggregations. We’ve created a script to create all of the trunks for you. Just issue the following commands. The last command assures that you don’t accidentally run the script a second time which would cause all sorts of issues. Feel free to review the code if you want to learn how to create trunks in FreePBX from the command line.

cd /root
wget http://incrediblepbx.com/add-skyetel
chmod +x add-skyetel
# uncomment next line if your incoming calls all have 10-digit numbers
# sed -i 's|from-trunk|from-pstn-e164-us|' add-skyetel
./add-skyetel
chmod -x add-skyetel

Incredible PBX Inbound Routing for Skyetel

Next we need to tell your PBX how to route incoming calls from Skyetel. Using a browser, log into the IP address of your PBX using your admin credentials. Because there is no trunk registration with Skyetel trunks, you will need to create an Inbound Route for every Skyetel DID. You cannot rely upon a Default inbound route because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes -> Add Inbound Route. For both the Description and DID fields, enter the 11-digit phone number beginning with a 1. Set the Destination for the incoming DID as desired and click Submit. Reload the Dialplan when prompted. Place a test call to each of your DIDs after configuring the Inbound Routes.

Incredible PBX Outbound Routing to Skyetel

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose Connectivity -> Outbound Routes -> Add Outbound Route. For the setup, we recommend the following using the CallerID Number you wish to associate with your outbound calls through Skyetel:

Enter the Dial Patterns under the Dial Patterns tab before saving your outbound route. Here’s what you would enter for 10-digit and 11-digit dialing. If you want to require a dialing prefix to use the Skyetel Outbound Route, enter it in the Prefix field for both dial strings.

Audio Issues with Skyetel

If you experience one-way or no audio on some calls, make sure you have filled in the NAT Settings section in the GUI under Settings -> Asterisk SIP Settings -> General. In addition to adding your external and internal IP addresses there, be sure to add your external IP address in /etc/asterisk/sip_general_custom.conf like the following example and restart Asterisk:

externip=xxx.xxx.xxx.xxx

If you’re using PJSIP trunks or extensions on your PBX, implement this fix as well.

Receiving SMS Messages Through Skyetel

Most Skyetel DIDs support SMS messaging. Once you have purchased one or more DIDs, you can edit each number and, under the SMS &MMS tab, you can redirect incoming SMS messages to an email or SMS destination of your choice using the following example:



Sending SMS Messages Through Skyetel

We’ve created a simple script that will let you send SMS messages from the Linux CLI using your Skyetel DIDs. In order to send SMS messages, you first will need to create a SID key and password in the Skyetel portal. From the Settings icon, choose API Keys -> Create. Once the credentials appear, copy both your SID and Password. Then click SAVE.

Next, from the Linux CLI, issue the following commands to download the sms-skyetel script into in your /root folder. Then edit the file and insert your SID, secret, and DID credentials in the fields at the top of the script. Save the file, and you’re all set.

cd /root
wget http://incrediblepbx.com/sms-skyetel
chmod +x sms-skyetel
nano -w sms-skyetel

To send an SMS message, use the following syntax where 18005551212 is the 11-digit SMS destination: sms-skyetel 18005551212 "Some message"

SMS and MMS Messaging with Postcards

Skyetel now has released a terrific, open source Docker app, Postcards, that lets you build an SMS and MMS messaging platform for your entire organization. Suffice it to say, anything you ever wanted to do with SMS and MMS messaging, you can do with Postcards. We won’t repeat Skyetel’s excellent tutorial, but you certainly need to visit their site and take Postcards for a spin.

NEW: Skyetel Support for Dynamic IP Addresses

You asked for it, and Skyetel has delivered. For Nerd Vittles users running servers with dynamic IP addresses, Skyetel now provides support for your platform. Log into your server as root and cd /usr/src. Then review this tutorial which describes the steps to put the pieces in place. Be advised that this is beta software at this juncture. If you run into issues, please post your questions on the PIAF Forum. Here are the actual steps:

(1) Log in to your Skyetel portal and Add a New Endpoint Group for your server giving it the name and current public IP address of your server.

(2) While still logged in, tap the Gear icon to open Settings dialog and choose API Keys tab.

(3) Add a new API key and write down your new SID and SID password.

(4) If your server is behind a router or firewall, log into that device and map UDP 5060 and UDP 10000-20000 to the private LAN address of your server.

NOTE: If your server is on the Debian, Ubuntu, or Raspbian platform, substitute the following command for the first two yum commands in step #5 below:

apt-get -y install coreutils curl git jq

(5) Log into your server and issue the following commands to install the EndPoint Updater:

yum -y install coreutils curl git epel-release
yum -y --enablerepo=epel install jq
cd /usr/src
git clone https://bitbucket.org/skyetel/ip-endpoint-group-update.git
cd ip-endpoint-group-update
./ip-update-endpointgroup.sh

(6) Fill in your credentials when prompted, and the cron script will be installed to keep your server’s dynamic IP address registered with Skyetel.

Introducing Skyetel’s New Fax Platform

Every time we read an article predicting the demise of fax technology, we have to chuckle. We’ve been reading the articles for about 30 years now, and fax still is the goto solution for many organizations. Can you spell HIPPA? Finally, Skyetel has dipped its toes in the fax waters by offering an easy-to-use fax solution for receipt of traditional and T.38 faxes. Simply purchase a Skyetel DID and configure it for vFax routing. Enter an email address for delivery of the faxes, and you’re done.


Sending faxes from the Skyetel portal still is on the drawing boards, but it’s coming. In the meantime, Incredible Fax™ which is bundled with all Incredible PBX® platforms will let you send faxes ’til the cows come home with our easy-to-use Hylafax/AvantFax implementation.

Implementing the New Spam Call Filter

One of the most often requested features for any PBX is spam call filtering. Skyetel takes it to the next level by dealing with the spammers before the calls ever reach your PBX. For each of your Skyetel phone numbers, click on the Features tab and set the Spam Call Filter as desired.

Recording and Transcribing Skyetel Calls

As with spam call filtering, recording and/or transcribing Skyetel calls is only a click away. For each of your Skyetel phone numbers, click on the Features tab and set the option desired for Recording and/or Transcribing calls. Recordings and Transcriptions can be managed from your Skyetel Dashboard. Storage is free for up to 30 days, after which they are deleted.

Skyetel Monitoring of Endpoint Health

In addition to monitoring and reporting the health of all Skyetel services in your web portal, this latest addition allows you to configure Skyetel to not only monitor the State of every registered endpoint but also its Health with realtime metrics of the Latency, Packet Loss, and Jitter of each of your endpoints. Simply check the Network QOS options desired.

Skyetel Expansion for Canadian Users


Here’s some great news for our Canadian friends. Skyetel has been listening!

  • Porting to Skyetel in Canada now is significantly easier and faster
  • Awesome reductions in audio round trip times
  • Epic reductions in time-to-deliver
  • Faster response times to technical issues (and fewer of them!)
  • Audio for Canadian calls will now originate from Canadian data centers
  • SMS and MMS available on Canadian ported numbers

Originally published: Thursday, November 1, 2018  Updated: Wednesday, June 12, 2019


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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Celebrating 2019: Return of the One-Minute Desktop PBX




If you’re new to the VoIP world and aren’t quite ready to dive into the Nerd Vittles cloud computing offerings, then we have a one minute setup solution today that doesn’t require you to buy anything ever. You can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® in less than 60 seconds. If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download the new Incredible PBX vbox image from SourceForge, double-click on the downloaded image, check the initialize MAC address box, and boom. In less than a minute, your PBX is ready to use.

The really nice thing about playing along today is it won’t cost you a dime to try things out for yourself. And, if you really love it and we think you will, there’s no hidden fee or crippleware to hinder your continued use of Incredible PBX for as long as you like. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk® that will revolutionize your communications platform. Just add your credentials and speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your nearest SIP phone. If you later decide you’d like to migrate your server to an inexpensive cloud-based platform, Incredible Backup and Restore make it a 15-minute turnkey task.

Installing Oracle VM VirtualBox

Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.

Installing Incredible PBX 13 with VirtualBox

To begin, download the latest Incredible PBX vbox image (2.6 GB) onto your desktop. Incredible PBX 13-13.10 includes all of the very latest FreePBX® 13 modules.

Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image and then click Import. Once the import is finished, you’ll see a new Incredible PBX virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.

(1) Click once on the Incredible PBX virtual machine in the VM List. Then (2) click the Settings button. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for Incredible PBX.

Running Incredible PBX in VirtualBox

Once you’ve imported and configured the Incredible PBX Virtual Machine, you’re ready to go. Highlight the Incredible PBX virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VM window. Always gracefully halt Incredible PBX just as you would on any computer.

Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your passwords immediately by typing: /root/update-passwords.

Setting the Date and Time with VirtualBox

On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Typing date will tell you whether your VM is affected. If it’s a problem, manually set the date and time and then update the hardware clock. Here’s how assuming 01070709 is the month, day, and correct time of your server:

date 01070709
clock -w

Overview of the Initial Asterisk Setup Process

For those new to PBXs, here’s a two paragraph summary of how Voice over IP (VoIP) works. Phones connected to your PBX are registered with Extensions so that they can make and receive calls. When a PBX user picks up a phone and dials a number, an Outbound Route tells the PBX which Trunk to use to place the call based upon established dialing rules. Unless the dialed number is a local extension, a Trunk registered with some service provider accepts the call, and the PBX sends the call to that provider. The provider then routes the call to its destination where the recipient’s phone rings to announce the incoming call. When the recipient picks up the phone, the conversation begins.

Looking at things from the other end, when a caller somewhere in the world wishes to reach you, the caller picks up a telephone and dials a number known as a DID that is assigned to you by a provider with whom you have established service. When the provider receives the call to your DID, it routes the call to your PBX based upon destination information you established with the provider. Your PBX receives the call with information identifying the DID of the call as well as the CallerID name and number of the caller. An Inbound Route on your PBX then determines where to send the call based upon that DID and CallerID information. Typically, a call is routed to an Extension, a group of Extensions known as a Ring Group, or an IVR or AutoAttendant giving the caller choices on routing the call to the desired destination. Once the call is routed to an Extension, the PBX rings the phone registered to that Extension. When you pick up the phone, the conversation begins.

Configuring Asterisk to Support NAT-Based Routing

With a VoIP server, many PBXs and Extensions are housed behind a NAT-based router that is found in most homes and businesses. These routers assign private IP addresses that are not accessible from the Internet. This causes SIP routing headaches because there are actually two legs to every call, one on the private IP address of your server or extension and another on the public Internet with an entirely different IP address. Routers supposedly handle this handoff of the call using Network Address Translation (NAT) and SIP ALG. With Asterisk-based PBXs, we want the PBX itself to handle the NAT chores so it is critically important to do three things when setting up your PBX. First, turn off SIP ALG on every router used by your PBX and every extension connected to your PBX. Second, tell your PBX about your public and private IP address setup. Step #2 is done in the Incredible PBX GUI with a browser. Login as admin and choose Settings:Asterisk SIP Settings. In the NAT Settings section of the form, click Detect Network Settings. Make sure your public and private IP addresses are correctly listed. Then click Submit and reload your dialplan when prompted. Failure to perform BOTH of these steps typically results in calls with one-way audio, i.e. where either you or the called party can’t hear the other party in the conversation. The third rule to remember is to always configure SIP Extensions on your PBX with NAT Mode=YES. This is rarely harmful and failure to configure SIP extensions in this way typically causes one-way audio in calls as well. IAX extensions avoid NAT issues.

Configuring Extensions with Incredible PBX GUI

Extensions are created using the Incredible PBX GUI: Applications:Extensions. Many SIP phones expect extensions to communicate on UDP port 5060. If this is the case with your SIP phone or softphone, then always create Chan_SIP extensions which communicate on UDP 5060. If your SIP phone or softphone provide port flexibility, then you have a choice in the type of SIP extension to create: Chan_SIP or the more versatile PJSIP. Just remember to always configure SIP extensions with NAT Mode=YES in the Advanced tab. If your VoIP phones or softphones support IAX connectivity, you may wish to consider IAX extensions which avoid NAT problems.

When you create a new Extension, a new entry is automatically created in the PBX Internal Directory. If you wish to allow individual users to manage their extensions or use the WebRTC softphone, then you will also have to create a (very) secure password for User Control Panel (UCP) access. Choose Admin:User Management and click on the key icon of the desired extension to assign a password for UCP and WebRTC access.

Configuring SIP Phones with Incredible PBX GUI

SIP phones and softphones typically require three pieces of information: the IP address of your server, the extension number, and the extension password. If you’re using a PJSIP extension, you also will need to change the port to UDP 5061. If your server is behind a NAT-based router, SIP phones also behind the same router need to use the private LAN address rather than the public IP address. If the SIP phones are outside the router protecting the PBX, then use the public IP address and make certain that you also map ports 5060 and 5061 from your router to the private LAN address of your PBX. Beginning with Incredible PBX 13-13.10, you now can make free SIP URI calls worldwide from almost any SIP phone or softphone. Our SIP URI tutorial covers everything you need to know.

The PIAF Forum can provide you with helpful information in choosing high quality SIP phones. Yealink phones are highly recommended with minimal issues. Cisco phones are the most difficult to configure. Insofar as free softphones, we recommend the Zoiper 3 offerings for Windows, Mac, iOS, and Android. Zoiper 5 still is experiencing some growing pains. A key advantage of the Zoiper softphone is it supports IAX extensions which eliminate the NAT issues entirely. On the Mac platform, we also recommend the Telephone app which is available in the App Store. For SRTP communications, use Grandstream Wave.

Configuring Trunks with Incredible PBX GUI

Perhaps the most difficult component to configure in the PBX is the Trunk. Almost every provider has a different way of doing things. We’ve taken some of the torture out of the exercise by providing configuration settings for dozens of providers. All you need to do is edit the desired Trunk (Connectivity:Trunks), change the Disable Trunk entry to No, and insert your credentials in both the PEER Details and Registration string of the SIP Settings Outgoing and Incoming tabs.

UPDATE: Whether your desktop PBX has a static IP address on the Internet or not, you now can take advantage of a terrific Nerd Vittles Skyetel offer of $50 in free service using Skyetel’s just released support for dynamic IP addressing. Start by mapping UDP ports 5060 and 10000-20000 to your server from your router. The firewall settings and Skyetel trunk setups are preconfigured in this VirtualBox image. Once you get this far, you’re ready to install Skyetel’s new dynamic IP address updater. This is required since you never actually register a trunk with Skyetel. Here’s how. Log into your server as root and cd /usr/src. Then follow this tutorial to put the pieces in place. While this is beta software at this juncture, we have tested it with excellent results. However, if you run into issues, please post your questions on the PIAF Forum. Now jump over to our Skyetel Tutorial to claim your $50 credit and to get your account set up and configured. Effective 10/1/2023, $25/month minimum spend required.

Of course, Incredible PBX comes preconfigured with setups for dozens of other providers that let you register a new trunk on the provider’s server. VoIP.ms (free iNUM), CircleNet, CallCentric (free DID and iNUM), LocalPhone (25¢/mo. iNUM), Future-Nine, AnveoDirect, and V1VoIP are excellent options.2 Most don’t cost you anything unless you make calls. Review our complete SIP tutorial here: Developing a Cost-Effective SIP Strategy.

Configuring Inbound Routes in Incredible PBX GUI

Inbound Routes, as the name implies, are used to direct incoming calls to a specific destination. That destination could be an extension, a ring group, an IVR or AutoAttendant, or even a conference or DISA extension to place outbound calls (hopefully with a very secure password). Inbound Routes can be identified by DID, CallerID number, or both. To create Inbound Routes, choose Connectivity:Inbound Routes and then click Add Inbound Route. Provide at least a Description for the route, a DID to be matched, and the Destination for the incoming calls that match. If you only want certain callers to be able to reach certain extensions, add a CallerID number to your matching criteria. You can add Call Recording and CallerID CNAM Lookups under the Other tab.

Configuring Outbound Routes in Incredible PBX GUI

Outbound Routes serve a couple of purposes. First, they assure that calls placed by users of your PBX are routed out through an appropriate trunk to reach their destination in the least costly manner. Second, they serve as a security mechanism by either blocking or restricting certain calls by requiring a PIN to complete the calls. For example, if you only permit 10-digit calls and route all of those calls out through a specific trunk with a $20 account balance, there is little risk of running up an exorbitant phone bill because of unauthorized calls unless you’ve deposited a lot of money in your account or activated automatic funds replenishment. This raises another important security tip. Never authorize recurring charges on credit cards registered with your VoIP providers and, if possible, place pricing limits on calls with your providers. If a bad guy were to break into your PBX, you don’t want to give the intruder a blank check to make unauthorized calls. And you certainly don’t want to join the $100,000 Phone Bill Club.

To create outbound routes in the Incredible PBX GUI, navigate to Connectivity:Outbound Routes and click Add Outbound Route. In the Route Settings tab, give the Outbound Route a name and choose one or more trunks to use for the outbound calls. In the Dial Patterns tab, specify the dial strings that must be matched to use this Outbound Route. NXXNXXXXXX would require only 10-digit numbers with the first and fourth digits being a number between 2 and 9. Note that Outbound Routes are searched from the top entry to the bottom until there is a match. Make certain that you order your routes correctly and then place test calls watching the Asterisk CLI to make sure the calls are routed as you intended.

Design Methodology for Outbound Routes

There are a million ways to design outbound calling schemes on PBXs with multiple trunks. One of the simplest ways is to use no dial prefix for the primary trunk and then use dialing prefixes such as *1 and *2 for the remaining trunks.

Another outbound calling scheme would be to assign specific DIDs to individual extensions on your PBX. Here you could use NXXNXXXXXX with the 1 Prepend as the Dial Pattern with every Outbound Route and change the Extension Number in the CallerID field of the Dial Pattern. With this setup, you’d need a separate Outbound Route for each group of extensions using a specific trunk on your PBX. Additional dial patterns can be added for each extension designated for a particular trunk. A lower priority Outbound Route then could be added without a CallerID entry to cover extensions that weren’t restricted or specified.

HINT: Keep in mind that Outbound Routes are processed by FreePBX in top-down order. The first route with a matching dial pattern is the trunk that is selected to place the outbound call. No other outbound routes are ever used even if the call fails or the trunk is unavailable. To avoid failed calls, consider adding additional trunks to the Trunk Sequence in every outbound route. In summary, if you have multiple routes with the exact same dial pattern, then the match nearest to the top of the Outbound Route list wins. You can rearrange the order of the outbound routes by dragging them into any sequence desired.

Configuring Incredible PBX for VirtualBox

In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.

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

This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.

To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.

In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂

All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.

Using Asteridex with Incredible PBX

AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.

Taking Incredible PBX for a Test Drive

You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.

With Allison’s Demo IVR, you can choose from the following options:

  • 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 Incredible PBX

News Flash: Turn Incredible PBX into a Fault-Tolerant HA Platform for $1/Month

Continue Reading: Configuring Extensions, Trunks & Routes

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

Originally published: Monday, January 7, 2019  Updated: Sunday, January 20, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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

A Sobering Look at Asterisk and the 2019 VoIP Landscape




Every six months or so we like to gaze into our crystal ball for a quick look at the VoIP landscape. 2018 has been quite the transformative year with the acquisition of Digium® and Asterisk® by Sangoma®. Unfortunately, as we predicted, the Digium layoffs have already begun, and 2019 may only get worse. While we have no inside information, we wouldn’t be surprised to see Digium’s headquarters in Huntsville closed within six months in an effort to balance the books. Part of the problem may be attributable to the terms of the purchase itself. However, we sense there’s a more troubling development. And that is the reality that VoIP is becoming less and less appealing to home users and small businesses as more and more folks migrate purely to cell phones. Those with teenagers already know this transformation is underway. With services such as Google Fi starting at $20 for unlimited calling and texting, it’s difficult to justify VoIP services even at bargain basement prices. Making the cellular switch even more appealing are offers such as a $400 credit with the purchase of an LG G7 smartphone from Google or a free LG G7 with new Sprint service.

What you lose with a pure cellular platform are many of the features that have made PBXs popular in the VoIP space: call routing, text-to-speech and voice recognition applications, conferencing, SPAM call blocking, and much more. But 2018 also was the year that Google finally pulled the plug on free calling through your PBX. Instead, you now have to purchase and configure a $50 OBi200 to continue with Google Voice, and the integration is painful to put it charitably. The demise of Google Voice added one more nail to the free VoIP coffin. And, as many of you know, Vitelity, our long-time platinum sponsor, now has bowed out of the VoIP retail business due to a change in focus from Voyant, the company’s new owner. Finally, our bargain-basement cloud provider for experimentation, HiFormance, appears to have bitten the dust. Details here. Suggestions here. Reminder: "You get what you pay for."

It’s not all bad news for 2019. First, all of the Incredible PBX platforms are still alive and well. And they will remain open source GPL code. Second, we’ve found a terrific new VoIP provider, Skyetel, that will give you a $50 credit so you can kick the tires for a good long while. Effective 10/1/2023, $25/month minimum spend required. Third, if you’re looking for a robust Cloud platform, Digital Ocean still is offering a $100 signup credit for your first 60 days of service, and Incredible PBX runs swimmingly on their $5/month platform with CentOS. Spend another $1 a month, and you get automatic backups of your cloud-based server. It’s cheap insurance for something as important as your phone system.

If you’re like us, you may be getting a little nervous about the future of Asterisk. We’ve already provided a series of articles on FusionPBX for FreeSWITCH. Our original tutorial and the follow-on articles showing how to create voice prompts using IBM Watson and how to create and deploy TTS applications such as news and weather reports are worth a careful read. And, if you consider yourself a pioneer, then you owe it to yourself to try out the FreeSWITCH developers’ new cloud-based platform, SignalWire. Here’s the $55 Promo code that worked for us: ITEXPO2019. That should get you off to a great start. And check out the pricing: U.S. DIDs are $0.08 per month, U.S. Origination rate (incoming) is $0.00325 per minute, U.S. Termination rate (outgoing) is $0.0072 per minute, U.S. SMS Outbound is $0.0009 per message, and U.S. SMS Inbound messages are free. MMS also available. Once verified, you can spoof any CallerID name and number that you own! What’s not to like? Asterisk Trunk setup example available here.

CAUTIONARY NOTE: SignalWire should be considered EXPERIMENTAL SOFTWARE and is not yet suitable for production use.

That should be enough excitement to keep all of you entertained over the holidays. We’re planning a few days off to be with family and friends. Let us be the first to wish each of you a very Merry Christmas. We’re looking forward to an exciting 2019!

Originally published: Monday, December 17, 2018



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



Spam Phone Call Blocker and CNAM Caching for FreePBX




Blocking spam phone calls has been a challenge to put it charitably. Thanks to some earlier work by Stewart Nelson on the DSLR forum as well as Stewart’s considerable hand-holding in the development of today’s tutorial, we want to introduce a new approach to blocking these calls. The way it works is first time callers that pass the TrueCNAM SPAM check will be prompted to "press 5 to connect." Since most spam calls sit in a queue for several seconds before a live person chimes in, that person won’t hear the prompt. After 10 seconds or an invalid response, a SIT tone is played and the call is disconnected. If you’d prefer, you can send the failed calls to voicemail by uncommenting a single line in your dialplan. When a successful caller calls again, the caller will be connected without encountering the press 5 prompt.1 While today’s approach won’t block every robocaller, our testing suggests that, in combination with TrueCNAM, it will catch more than 95% of the spam callers. Using CallerID Superfecta with CNAM lookups from OpenCNAM coupled with AsteriDex and the Asterisk® Phonebook will provide an extremely low-cost solution both for blocking spammers AND for displaying accurate CNAM data for incoming calls since you’ll only pay for CNAM and TrueCNAM lookups from legitimate callers once.

Here’s the actual dialplan addition that will monitor your incoming calls:

[sub-log-caller]
exten => s,1,NoOp(*** begin sub-log-caller ***)
exten => s,n,GotoIf(${DB_EXISTS(cidname/${CALLERID(num)})}?CNAMCHECK)
exten => s,n,GotoIf($[${DB_EXISTS(SPAMCHECK/deactivate)} = 0]?ACTIVATE)
exten => s,n,GotoIf($[${DB(SPAMCHECK/deactivate)} = 1]?CONNECTNOW)
exten => s,n(ACTIVATE),NoOp(Not yet WhiteListed)
exten => s,n,AGI(truecnam.sh,${CALLERID(number)})
exten => s,n,GotoIf($["${SPAM}"="SPAM"]?FLUNKED)
exten => s,n,Playback(silence/1)
exten => s,n,Playback(to-call-num-press)
exten => s,n,Playback(digits/5)
exten => s,n,Read(MYCODE,beep,1,n,1,10)
exten => s,n,GotoIf($["${MYCODE}" = "5"]?ANONTEST)
exten => s,n(FLUNKED),NoOp(*** Caller FLUNKED screening ***)
;exten => s,n,Dial(local/*701@from-internal) ; uncomment to send to 701 VM
exten => s,n,Zapateller()
exten => s,n,Hangup
exten => s,n,Return()
exten => s,n(ANONTEST),GotoIf($[${CALLERID(num)} > 0]?WHITELIST:CONNECTNOW) 
exten => s,n(CNAMCHECK),Set(CNAM1=${CALLERID(name)})
exten => s,n,Set(CNAM2=${DB(cidname/${CALLERID(number)})})
exten => s,n,GotoIf($["${CNAM1}" = "${CNAM2}"]?WHITELISTED
exten => s,n(WHITELIST),Set(DB(cidname/${CALLERID(number)})=${CALLERID(name)})
exten => s,n,Set(CALLERID(all)="${CALLERID(name)} < ${CALLERID(number)}>")
exten => s,n(WHITELISTED),NoOp(WhiteListed: ${CALLERID(all)})
exten => s,n(CONNECTNOW),NoOp(*** end of sub-log-caller ***)
exten => s,n,Return()

We first introduced some of the CallerID caching concepts in our previous article last May. That article also documented the procedure for adding inbound call processing logic into FreePBX. If you already have implemented the steps outlined in that article, then the only modification required to deploy today’s new spam blocking technique is to replace the [sub-log-caller] context and reload the Asterisk dialplan. NOTE: Some deployments of CallerID Superfecta have an incorrect database password in the Default setup for AsteriDex. The original article will walk you through making the necessary change.

If you’re starting from scratch, stop here for a bit and follow all of the steps in our previous article which now incorporates the spam blocking code as well. Here’s the link to get started. Return here once you’ve completed the initial setup.

If you’re updating a previous deployment, here are the steps. Edit extensions_custom.conf in /etc/asterisk and remove the [sub-log-caller] context toward the end of the file. Then save the file. Next, issue the following commands to move the TrueCNAM script into place and insert the updated [sub-log-caller] context as well as the new [macro-dialout-trunk-predial-hook] context. Then reload your Asterisk dialplan. The dialplan additions will populate the Asterisk Phonebook and also whitelist calls from your PBX as well as incoming calls making it through the Spam Blocker.

cd /tmp
wget http://incrediblepbx.com/sub-log-caller.tar.gz
tar zxvf sub-log-caller.tar.gz
rm -f sub-log-caller.tar.gz
mv truecnam.sh /var/lib/asterisk/agi-bin
cd /etc/asterisk
cat /tmp/sub-log-caller.txt >> extensions_custom.conf
asterisk -rx "dialplan reload"

 

Rotary Dial Phones & Blocked Numbers

If someone you know and love still has a rotary dial phone, then you will need to manually add their number to either AsteriDex or your Asterisk Phonebook. Otherwise, the calls will never make it through the Spam Catcher. You can do this within the FreePBX GUI by accessing Admin -> Asterisk Phonebook. Click + Add Phonebook Entry and enter the 10-digit number for Grandma as well as her name. Add a second entry with Grandma’s 11-digit number in case some of your VoIP providers happen to send 11-digit CallerID numbers. We hasten to add you should normalize the formatting of your CallerID numbers as quickly as you can to avoid double entries. For those in the U.S. and Canada, we recommend the from-pstn-e164-us context for all of your trunks.

If you have lots of friends with rotary dial phones or if you get calls from important, but unknown numbers such as medical offices where Caller ID numbers are blocked, then you probably should consider uncommenting the voicemail option in [sub-log-caller]. Then you at least will get voicemail notifications when one of these callers attempts to contact you. You still will have to manually add them to AsteriDex or the Asterisk Phonebook so they can contact you directly in the future. HINT: Most medical office calls now spoof the main number of the office so you only need to add the office number just as you did with grandma.

Toggling Spam Blocker On and Off

We’ve also included the ability to turn off the Spam Blocker should you ever wish to do so. To disable the Spam Blocker, issue the following command at the Asterisk CLI:

database put SPAMCHECK deactivate 1

To once again enable the Spam Blocker, issue the following command at the Asterisk CLI:

database deltree SPAMCHECK

WhiteListing Previous Callers

We appreciate that you may not want to aggravate callers that have been calling you for years by making them jump through hoops the next time they call. So here’s a quick way to populate your Asterisk Phonebook with the names and numbers of previous callers. For entries where the CNAM is merely the CallerID Number, future calls from these numbers still will be looked up with OpenCNAM to obtain an actual CNAM match. We’ve made a couple of assumptions that you are more than welcome to adjust to meet your own needs. First, we’ve limited the list to callers from the past two calendar years. Second, we’ve only captured calls that lasted more than 15 seconds. We’ll drop down to the Linux CLI to build the list of callers to import. Then we’ll use the FreePBX GUI to import the list into the Asterisk Phonebook. While we’re building the import list, you’ll have two opportunities to prune the list using your favorite text editor. To get started, issue the following commands from the Linux CLI:

mysql -u root -ppassw0rd asteriskcdrdb -Ns -e "select distinct src, clid \\
from cdr where calldate > '2017/01/01' and duration > 15 \\
order by clid asc" > 2YR-full

Now edit the 2YR-full file and remove any complete lines you don’t want to import.

Next, we’ll reformat the CallerID Numbers and Names into a format needed for the import:

cat 2YR-full | cut -f 1 -d '"' | sed 's|[[:space:]]||' > 2YR-numbers
cat 2YR-full | cut -f 2 -d '"' > 2YR-names
paste 2YR-numbers 2YR-names | awk '{print $1,$2,$3,$4}' > 2YR-all
awk '{print $2 " " $3 $4 ";" $1";"}' 2YR-all > 2YR-freepbx.csv

Now we should have our 2YR=freepbx.csv file in its final form for import. Open the file in your favorite editor. The syntax of the entries should be CallerID Name, then a semicolon, then CallerID Number, and then a semicolon. Discard any additional lines you wish to exclude from the import. Once you have all the entries squared away, copy the file to your desktop PC and open FreePBX in your browser. Navigate to Admin -> Asterisk Phonebook. Click Import Phonebook and then Browse. Select the 2YR-freepbx.csv file from your desktop. Then click Upload. Take a final look at the new entries in your Asterisk Phonebook to make sure nothing came unglued, and you’re all set.

TrueCNAM: The Icing on the Spam Cake



A couple years ago we introduced TrueCNAM, a service that provides not only CNAM data but also Caller Reputation scoring. Those that flunk using the revolving caller reputation matrix get disconnected automatically. We strongly encourage you to add the TrueCNAM service to your PBX. The service includes a free tier as well as incredibly reasonable commercial tiers. For background on the service, here’s a link to our previous TrueCNAM tutorial. For today, start by signing up for a TrueCNAM account and obtain an APIkey and APIpassword. Then register at least one of your DIDs with the service. Once you have your credentials and your DID number in hand, edit truecnam.sh in /var/lib/asterisk/agi-bin. Insert these three items at the top of the file and save it to activate TrueCNAM. It doesn’t get much easier than that.

Now make a few test calls to your PBX to assure that everything is working as documented. Enjoy!

Originally published: Monday, November 26, 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.



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. Once installed, you can change the voice prompt to a number other than 5 by modifying lines 10 and 12 of the context sub-log-caller which you will find in extensions_custom.conf in the /etc/asterisk directory at the completion of this install. []

R.I.P. GVSIP: A Final Farewell to Google Voice



It’s been a death by a thousand cuts, but today marks the end of the Google Voice era with Asterisk®. Since Google removed XMPP support and transitioned to their new GVSIP platform, many have held out hope that Google hadn’t moved to a purely commercial platform with their ObiHai deal. Yesterday, the head of the Google Voice project requested that all Asterisk GVSIP implementations be discontinued citing Google’s Terms of Service. We hinted this was coming back in July and have reproduced our tweet below. We have since removed all of our articles pertaining to GVSIP, and we would encourage all of our readers to honor Google’s wishes and move on. We’ve made it easy with a $50 gift certificate from Skyetel (expires March 31, 2019). It will buy you many months of free VoIP service.



You still have several options with your Google Voice trunks. First, you can forward all incoming calls to Google Voice to another phone or DID of your choosing. This costs you nothing other than a minute to set it up. Second, you can port out your Google Voice number to another provider. Skyetel will cover your porting expense at their end during your first 60 days of service. Google charges $3 to port out your number unless you originally ported it into Google in which case it is free. Here’s how. Although we’re not big fans, a third option is to purchase an OBi200 device and continue to use your Google Voice trunk with Asterisk. Our tutorial from last May will show you how. Effective 10/1/2023, $25/month minimum spend at Skyetel is required.

As we’ve mentioned often, the beauty of VoIP is not having to put all of your telephony eggs in a single basket. Google’s latest move reinforces how important it actually is to configure several VoIP trunks on your server. While Skyetel and Vitelity are both excellent primary trunks and rarely experience an outage, it’s still a good idea to have a backup. VoIP.ms (free iNUM), CircleNet, CallCentric ($1/mo. DID and iNUM), LocalPhone (25¢/mo. iNUM), Future-Nine, AnveoDirect, and V1VoIP are excellent options. Most don’t cost you anything unless you make calls. Review our complete SIP tutorial here: Developing a Cost-Effective SIP Strategy.

Dale Carnegie Award: ObiHai Man of the Year

Originally published: Friday, November 16, 2018



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.