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The Most Versatile VoIP Provider: FREE PORTING

Introducing OSS Endpoint Manager for FreePBX 16 & Incredible PBX 2027


If you’ve been searching for a free, open source provisioning system for your SIP telephones and you’re using Incredible PBX 2027 or other FreePBX® 16 platforms, your prayers have been answered thanks to the hard work of Bill Simon. What this buys you (for free) is a quick way to configure SIP devices from most of the major players shown above including Aastra, AudioCodes, Cisco, Linksys, CloudTC, Digium, Grandstream, Intelbras, Mitel, Norphonic, Patton, Polycom, Snom, Thomson, Unidata, Xorcom, and Yealink.

To get started, log into FreePBX on your PBX platform and click on the Settings tab. If you see OSS Endpoint Manager as you will on most Incredible PBX 2027 installations, you can skip installing OSS Endpoint Manager and move on to the next step. Otherwise, here are the steps to install OSS EPM 16. Using SSH, log into your PBX as root and issue the following commands:

cd /var/www/html/admin/modules
wget https://filedn.com/lBgbGypMOdDm8PWOoOiBR7j/OSS-EPM/ossepm16.tgz
tar zxvf ossepm16.tgz
fwconsole ma install endpointman
rm -f /tmp/*
fwconsole reload 

Once you have OSS Endpoint Manager installed, return to the FreePBX GUI and open the application under the Settings tab. Read the reminder that you’re not using Sangoma’s commercial endpoint management product. Then click on the pull-down box in the right margin. Now let’s get the templates for the various SIP devices loaded onto your server.

  1. On the right pull-down, choose EndPoint Manager Settings.
  2. Adjust the entries as desired to set up your server as the host.
  3. For Package Server, enter: https://ossepm.incrediblepbx.com/
  4. Click in any other field to save your entries.
  5. On the right pull-down, choose Template Manager.
  6. Click the displayed link to download the latest templates from GitHub.
  7. On the right pull-down, choose Package Manager.
  8. Click Check for Updates.
  9. Click Install button beside each desired Brand.


The rest is a walk in the park. Plug in your SIP devices and perform the manufacturer’s steps to place the device in configuration mode. Insert the credentials to access OSS EPM on your PBX. Then sit back and enjoy the show.

Originally published: Thursday, February 1, 2024



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.
 



Deal of the Year: OBi2182 Color WiFi Phones now $59.99



One year and counting… If you still have some Google Voice numbers lying around or if you’d like to join the party a little late, now is your second chance. Amazon now is selling the OBi2182 phones to U.S. residents with native Google Voice as well as WiFi, Bluetooth, HD Voice, speakerphone, and a power adapter currently selling for $69.99. Hold, transfer, and conferencing support are included in the phone. Here’s our Amazon referral link to keep the lights on.

We were big fans of OBiHai once upon a time. But then they sold out to Polycom. And they subsequently sold out to Plantronics. And support went from bad to worse along the way. But things appear to have stabilized so this offering may just be one of the best deals on the planet. Not only do you get unlimited free calling throughout the U.S., Canada, and Ukraine but you also get some of the best pricing ever on most international calls. While we no longer have any dealings with ObiHai, Polycom, or Plantronics, we know many of our followers were left high and dry when Google pulled the plug on Google Voice for Asterisk®. Here’s a tip.

Adding one or more of these phones makes a perfect supplement to your Asterisk platform making it easy to add free calling to your home or home office PBX. And it’s not too late to order a Google Voice number if you’re new to the game. Configuring a SIP or PJsip connection as well as Bluetooth connectivity to your cellphone is just as simple as adding Google Voice trunks using the OBiTalk web interface shown below. Trunk setups take less than a minute apiece. The phone supports 12 lines. And WiFi calls are crystal-clear thanks to HD Voice.


Grab an OBi2182 while you still can! An Admin Guide to the OBi2182 is available here.

Originally published: Monday, May 9, 2022



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.
 



5 Minute Wonder: Incredible PBX 2022 in Cloud for $25/Yr.



We’ve been building turnkey Incredible PBX® servers for virtual machine platforms for many years. Because the servers are built from images, typical install times have been 5 minutes or less with Proxmox, VirtualBox, and VMware ESXi. But the missing piece has been a similar image install on a true cloud-based platform. This limitation was primarily due to the fact that we don’t own or control the available cloud platforms which typically limit image installs to operating systems such as CentOS, Debian, Ubuntu, and Windows. However, CrownCloud was good enough to add our Incredible PBX 2021 Debian image and the Incredible PBX 2020 CentOS 7 image to their portfolio. And, today, we have more good news. CrownCloud has now added the Incredible PBX 2022 image for Rocky 8 as well.

These 5-minute turnkey installs of Incredible PBX 2020 for CentOS 7, Incredible PBX 2021 for Debian 10, and now Incredible PBX 2022 for Rocky 8 are being offered at the jaw-dropping price of $25 a year. The monthly cost is cheaper than a cup of coffee at Starbucks, and you’ll have a fully-functioning, production-ready KVM platform including a free snapshot with 1GB RAM, 20GB SSD storage, and 1TB of monthly bandwidth in your choice of server locations including Los Angeles and Atlanta in the United States as well as Germany and the Netherlands in Europe. And, unlike all of the other Asterisk® aggregations, Incredible PBX still provides a source code-based platform that can be tailored to meet any special requirements your organization may need.

We don’t make a nickel on these offerings so consider this our special thanks to all of our loyal fans.

Here are the links to sign up for the service and take advantage of these Incredible PBX deals:

When you sign up for the service, choose any traditional OS for the base install. Once it’s on line, go into CrownPanel and choose Reinstall. Then select Incredible PBX 2022 for Rocky from the Application Images pulldown as your new install. In less than 5 minutes, your server will be ready for you to login. Be sure to use SSH and NOT the VNC utility included in CrownPanel. This will assure that your desktop machine’s IP address gets whitelisted in the Incredible PBX firewall. Otherwise, you won’t be able to SSH into your server from your desktop. Once you login, the Incredible PBX configurator will prompt you to set passwords for root login, admin login to FreePBX, and admin login credentials for Apache to access AsteriDex and Reminders. Add a trunk provider (Skyetel is preconfigured and enabled by default) and a softphone or Incredible PBX SIP phone, and your PBX is fully operational. Check out the CrownCloud Wiki.


Planning Ahead for That Rainy Day

One of our favorite features of Crown Cloud is the free snapshot (a.k.a. backup) at no additional charge. We recommend you take snapshots regularly as you make major changes in your server’s configuration. In this way, if something comes unglued, you can easily restore the snapshot and never miss a beat. You’ll find the Remote Snapshot option in your CrownPanel menu.

Configuring Skyetel for Incredible PBX

If you’ve decided to go with Skyetel, here’s the drill. 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 the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

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

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: my.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. Everything is already in place on the Incredible PBX 2022 side of the house so you can start making and receiving calls immediately.

Configuring ClearlyIP SIP Trunking

For the tightest integration with FreePBX, no SIP provider holds a candle to Incredible PBX SIP Trunking with ClearlyIP. The reason is fairly obvious. The ClearlyIP folks were the original developers of FreePBX. In addition to all of the traditional SIP trunking services, you also get CNAM support and state-of-the-art E911 service which can be deployed in full compliance with Kari’s Law and the Ray Baum Act. If you’re a system integrator and don’t know about your financial liability for failure to comply with the new rules, it’s time to do some reading.

Configuring VoIP.ms for Incredible PBX

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

Configuring Anveo Direct for Incredible PBX

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

Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Enable the Trunk. Then click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup. In the Incredible PBX GUI, be sure to enable all of the remaining Anveo trunks.

By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:

cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"

Configuring a SIP Phone for Incredible PBX 2022

We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which is a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. We obviously recommend the Incredible PBX IP phones from ClearlyIP which are the most versatile.

If you’ve been keeping up with recent Nerd Vittles developments, then you already know that we have just introduced a new Cellular Phone which connects directly to your PBX and serves as a perfect remote extension and traveling companion. You can read all about it here.

Software devices such as the YateClient softphone are another option for desktop machines. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the VoIP-Info.org Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server plus :5061 for the PJsip 701 extension, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

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

DEMO - Apps Demo
123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
TODAY - Today in History
LENNY - The Telemarketer's Worst Nightmare

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store. For Android users, check out the terrific new VitalPBX Communicator. Works flawlessly with Incredible PBX.

For smartphone solutions, visit the Incredible PBX Wiki for our softphone recommendations.

Configuring SendMail with Incredible PBX

In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. We strongly recommend configuring SendMail using either your ISP or Gmail as an SMTP Relay Host. NOTE: If you are using a Gmail account with 2-step verification enabled, you MUST use a Gmail App Key instead of your Gmail account password. You also must enable Less Secure Apps access to the Gmail account.

Configuring a Gmail account with Incredible PBX 2022 is as simple as entering your Gmail credentials. Just run this script: /root/enable-gmail-smarthost-for-sendmail.

Here are the steps using a Gmail account with Incredible PBX 2020:

cd /etc/mail
yum -y install sendmail-cf
hostname -f > genericsdomain
touch genericstable
cd /usr/bin
rm -f makemap
ln -s ../sbin/makemap.sendmail makemap
cd /etc/mail
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
# Stop here and edit client-info (nano -w client-info) in all three lines.
# Replace  user_id with your gMail account name without @gmail.com
# Replace password with your real gMail password OR
#  use your Gmail App Key if 2-step verification is enabled
# Be sure to replace the double-quotes shown above if they don't appear in the file!!!
# Save your changes (Ctrl-X, Y, then Enter)
chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
systemctl restart sendmail

Even though these servers are hosted in the cloud, we still recommend using a SmartHost to minimize email delivery problems.

Test outbound mail using this command with your actual email address:

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

On some implementations, you may notice in the FreePBX GUI that the mail queue has failed. Here's the fix:

chmod 777 /var/spool/mqueue
service sendmail restart

Once you are sure your emails are being delivered reliably, here's a sample GUI voicemail configuration for an extension:



Be advised that Google has hinted that the Gmail Smarthost landscape may be changing. See our recent article for a simple SmartHost alternative.

Incredible PBX Administration

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

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

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

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

admin-pw-change is used to set the admin password for access to the FreePBX/Incredible PBX web GUI using a browser pointed to the local IP address of your server.

apache-pw-change is used to set the admin password for access to Apache/Incredible PBX apps including AsteriDex and Reminders. This provides a password layer of protection for access to these applications.

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

reset-extension-passwords is a script that automatically and randomly resets ALL of the SIP passwords for extensions 702-705. Be careful using this one, or you may disable existing registered phones and cause Fail2Ban to blacklist the IP addresses of those users. HINT: You can place a call to the Ring Group associated with all five extensions by dialing 777.

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

show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX actually is. A similar listing is available in the GUI at Admin -> Feature Codes.

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

ssh-regen.sh allows you to reset the SSH keys for your server for added security.

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

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

Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"

To remove call forwarding: asterisk -rx "database del CF 705"

Keeping FreePBX 15 Modules Current

We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

rm -f /tmp/*
fwconsole ma upgradeall
fwconsole reload
/root/sig-fix
/root/sig-fix

Where To Go From Here

Complete documentation on the ClearlyIP Devices Module is available here.

Complete documentation on the FreePBX GPL Modules is available here.

Complete documentation on the Incredible PBX additions is available here.

An introduction to configuring extensions, trunks, and routes is available here.

Free voicemail transcription with email delivery. Tutorial available here.

Setting Up a VPN for Your PBX: OpenVPN or NeoRouter
 

Originally published: Monday, March 28, 2022



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 OpenSIPS 3 for Incredible PBX and Debian 10


Today we’re pleased to introduce an updated OpenSIPS installer for Debian 10 featuring the latest release of OpenSIPS. Our previous tutorial with Debian 8 is now obsolete, an all-too-frequent occurrence in the open source world. Today’s open source SIP server lets you connect users to make and receive free as well as commercial calls worldwide. There’s excellent documentation making it easy to integrate into our existing Incredible PBX platform without hiring a consultant. It’s also straight-forward to secure without providing free phone service to every bad guy on the planet.

OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others. Source: opensips.org

We’ve often complained that the problem with many open source projects is that the developers get so focused on making money that they skimp on the documentation to encourage consulting work or participation in expensive conferences. We have found just the opposite with OpenSIPS. In fact, much of today’s implementation is based upon an excellent tutorial by the folks at PowerPBX. Down the road, if you find yourself in need of a consultant, their services would be a good place to start. What we’ve added to the PowerPBX design is security, support for clients behind NAT-based routers, and an integration scheme for Asterisk®, FreePBX®, and Incredible PBX® platforms so that you get the best of all worlds, a public facing SIP server with the UC feature set that most organizations expect. Last but not least, our turnkey GPLv2 installer will get you up and running in about 5 minutes.

Choosing an Appropriate Platform for OpenSIPS

Let’s begin by addressing the appropriate platform for an OpenSIPS server. The server needs to have a public IP address that is static, and the server should not be situated behind a NAT-based router. It only complicates things and is beyond the scope of what we plan to address. For those that are frequent visitors, you already know that we’ve been pushing everyone to kiss their local hardware goodbye and join the cloud revolution. When it comes to public-facing VoIP platforms like OpenSIPS, most of us don’t have a choice. You need a static IP address on the open Internet. And, for the sake of security, a KVM cloud platform is a must since older OpenVZ platforms don’t support the ipset component of IPtables which makes it easy to block hundreds of thousands of IP addresses without a performance hit on your server. Pure whitelist access simply isn’t an option if you wish to retain the functionality of a VoIP application such as OpenSIPS.

Ten to twenty gigabytes of disk space should be more than ample for OpenSIPS. The amount of RAM in your server depends upon the volume of calls your server will be handling. If it’s a dozen simultaneous calls then 1GB of RAM will suffice. If it’s 100,000 calls, then take a look at this article for tips on sizing your server. For today’s implementation, you’ll need a Debian 10 platform so a low-cost KVM provider including Digital Ocean, Vultr, and OVH should be fine.1

Choosing OpenSIPS Components to Deploy

We’ve divided up today’s tutorial into bite-sized pieces so that you can pick and choose where to stop implementing and start using. You do not need to have an Asterisk server to make and receive calls with OpenSIPS. However, OpenSIPS lacks voicemail and AutoAttendant/IVR components so, if those are a requirement, then you either need a VoIP service provider that offers them, or deploy a $50 Incredible PBX for the Raspberry Pi to add the missing pieces.

What OpenSIPS offers is a free server platform for worldwide SIP communications so that you, your friends, and business associates can call or connect from anywhere using freely available SIP softphones or any of dozens of SIP telephone instruments. We’ll stick with softphones for today, but hardware-based SIP telephones are equally simple to deploy.

This is not a criticism because it is one of the best tutorials we’ve ever used but, if you want to see how complex a typical OpenSIPS server deployment is, take a look at the PowerPBX tutorial we used as a starting point with OpenSIPS. We’ve compressed most of those procedures into a turnkey installer that only requires you to enter a MySQL root password of passw0rd (with a zero) once you have your Debian 10/64 platform up and running.

Deploying a Debian 10 Server Platform

Start by choosing a cloud provider that offers the 64-bit Debian 10 minimal platform as a deployment option. Most do. As noted, we recommend a KVM platform with support for ipset making it easy to block entire countries overrun with bad guys. Choose offerings with at least 1GB RAM and a 10GB drive to get started. Configure your Debian 10 server with a fully-qualified domain name (FQDN). This is critically important with our security design because we will assign all OpenSIPS users/extensions to this FQDN and reserve your server’s IP address purely for connections from service providers and Asterisk servers. This makes it all but impossible for anyone to hack into your server since most script kiddies launch attacks on IP addresses, not FQDNs. Using an unusual FQDN adds an extra layer of security, but that’s your call. If you lack the ability to assign FQDN aliases to a domain which you own, you can obtain a free FQDN from numerous sources including ChangeIP and point it to the IP address of your OpenSIPS server.

Installing OpenSIPS on a Debian 10 Server

Now the fun begins. Log into your Debian 8 server as root and issue the following commands to prepare for the OpenSIPS install:

cd /root
wget http://incrediblepbx.com/opensips3.tar.gz
tar zxvf opensips3.tar.gz
rm -f opensips3.tar.gz

Make sure you have logged into your Debian 10 server as root using SSH or Putty from a desktop PC that you will use to manage OpenSIPS with a browser. The reason is because this IP address automatically will be whitelisted in the OpenSIPS firewall as part of the install process. Otherwise, you will need to manually log into SSH and whitelist the IP address of your desktop PC using /root/add-ip each time you wish to access the OpenSIPS Control Panel since TCP port 80 (HTTP) is not exposed to the public Internet as a security precaution.

To begin the install, issue this command: /root/install

As the install progresses, you’ll first be prompted to choose the GRUB install device. Press the spacebar on the first entry. Then press TAB and ENTER. When prompted for the SSH configuration, choose "keep local version" and then press TAB and ENTER. For the MariaDB setup, press ENTER when prompted for the current password. Type N when prompted whether to switch to unix_socket authorization. Then type Y to change the root password. Be sure to use passw0rd (with a zero) as your MySQL password, or the install will fail. This is NOT a security risk unless your Debian 10 root user account is compromised. And, in that case, it won’t matter anyway since the MySQL password could easily be changed. Type Y to remove anonymous users. Type Y to disallow remote root logins. Type Y to reload the MySQL privilege tables.

Next you’ll be prompted to set your timezone and TZ entries. For East Coast U.S., it’s 2,49,1,1 then America/New_York. Later you’ll be prompted twice for the MySQL root password. You must enter passw0rd (with a zero). When the OpenSIPS status screen displays, type Q to exit the display. There are a couple of steps where you will be prompted for input. Correct responses are indicated before the various prompts. Pay particular attention when you are prompted to change the SSH port from TCP 22 to a port number in the 1000-2020 range as a security precaution. We recommend using the year you were born because it will be easy for you to remember. When the install finishes and you log out of your server, the next SSH login will look like this where XXXX is the SSH port you chose and yyy.yyy.yyy.yyy is the OpenSIPS server address: ssh -p XXXX root@yyy.yyy.yyy.yyy


Although most of the configuration of your OpenSIPS server will be handled using a web browser and the OpenSIPS Control Panel GUI, we’ve included a few scripts in /root to assist with maintenance of your server platform. Here’s a brief summary of the script functions:

  • pbxstatus – Status of your OpenSIPS server (image sample above)
  • add-ip – Temporarily WhiteList IP address until next iptables-restart
  • ban-ip – Permanently Ban an IP address
  • unban-ip – Unban a previously banned IP address
  • log-purge – Zero out all of the major Linux log files
  • opensips-check – Assures OpenSIPS and RTPproxy are running (runs automatically)
  • Fail2Ban BlackListsiptables -nL | grep -A100000 "opensips ("
  • IPset BlackList (KVM/OVZ7 platforms only) – ipset list | sort

We secure your server in several ways: (1) by disguising the SSH port, (2) by locking down almost every port on your server with the IPtables firewall with the exception of the SIP ports, (3) by deploying Fail2Ban to scan your OpenSIPS log for errors and lock out attackers for an extended period of time, and (4) by deploying the IPset blacklist for KVM platforms. With this design, there is a symbiotic relationship between IPtables, Fail2Ban, and IPset. Therefore, it is critically important that you only restart these services using the iptables-restart command. NEVER issue other IPtables commands to restart or save your firewall settings.

Activating a SIP Server with OpenSIPS Control Panel

We don’t want to overload you on the first day with your new OpenSIPS 3 platform so we’ll walk you through the preliminary setup steps to create your SIP Domain. Then we’ll show you how to set up user accounts (also known as extensions). Finally we’ll walk you through setting up a trunk to make and receive calls from a commercial SIP provider. When we’re finished today, you’ll be able to make and receive calls using SIP URIs or DIDs which you have purchased from a provider. Then next week we’ll focus on integration of OpenSIPS with an Asterisk platform of your choice using Incredible PBX as an example. Once we’re finished, you’ll be able to handle user account registrations exclusively on your OpenSIPS server while leaving your Asterisk platform completely hidden from public exposure.

Logging into the OpenSIPS Control Panel

As deployed, the OpenSIPS Control Panel is accessible via web browser. As noted previously, HTTP Port 80 access is blocked by default unless the IP address of your desktop PC has been whitelisted either as part of the initial install or using the add-ip script in /root. Once your desktop PC’s IP address is whitelisted, point your browser to http://xxx.xxx.xxx.xxx/cp



The default Username is admin, and the default password is opensips. Once you’re logged in, immediately click on the Users icon in the upper-right corner of the dashboard. Then click the Edit Info pencil icon for user Admin and change your password. Click Save when done.

Creating Domains with OpenSIPS Control Panel

In the Left column of the Dashboard, you’ll see two tabs: Users and System. Click on the System tab to expose the available choices. Then choose the Domains option.



Domains are the essential building blocks in OpenSIPS. You can manage one or a hundred domains on a single OpenSIPS server, and each domain can have its own set of Users, Trunks/Gateways, and Dialplan rules. We’re actually going to create two domains, one for the IP Address of your OpenSIPS server and a second one for the FQDN of your OpenSIPS server. For added security, we will create all User accounts under the FQDN Domain. And we’ll reserve the IP Address Domain for DID Trunks/Gateways from registered, commercial SIP providers. This design allows attackers to attempt to register to accounts on your IP Address Domain until the cows come home, and they will never be successful because there are no existing SIP user accounts there. Keep it that way! With our OpenSIPS design, Fail2Ban will block attackers after a single failed registration attempt. And OpenSIPS itself will identify and block all SIP flood attacks using either Fail2Ban or IPset.

Now that you understand the design, let’s set up your domains. After choosing System -> Domains, enter the IP Address of your OpenSIPS server at the SIP Domain prompt. Then click Add New Domain followed by Reload on Server. Repeat the same steps to enter the fully-qualified domain name (FQDN) of your OpenSIPS server. When finished, you should see:


Creating Users with OpenSIPS Control Panel

We’ve already explained the security implications and reason for creating User accounts with your FQDN Domain only. Click on Users -> User Management -> Add New to get started. You can use Numbers (what we call Extensions in Asterisk) or Names. Our preference is to use Numbers for the User accounts and then to create Alias Names (as desired) for each User account. You can’t dial names from most SIP telephones. This also keeps the design similar to what many are used to in the Asterisk environment. A completed dialog would look something like the following. Use the Domain pull-down to choose your FQDN. Obviously, the passwords must be secure and must match. Then the Register button will be enabled to save. The actual Numbers used for Usernames are completely up to you.



Create at least a couple User accounts so that you can set up two SIP phones to call yourself and verify that everything is working. These User accounts become an integral part of the SIP URI to receive calls from any SIP phone in the world: 7701@opensips.yourdomain.com

Before you can actually answer an incoming call to your SIP URI, you’ll need to register the User account using either a softphone or SIP phone. We’ll do that next. But, first, let’s create an Alias to 7701 User so that folks can reach you by calling joe@opensips.yourdomain.com

Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the example below. Make sure that you select your FQDN Domain using the pull-downs for BOTH the Domain and Alias Domain fields. Then click Add to save.


Registering a Softphone to an OpenSIPS User Account

There are literally dozens of free SIP soft phones from which to choose. We covered some of our favorites for every platform in previous articles. For our purposes today, we recommend you choose one of the Linphone softphones which are available for the PC, Mac, Linux, Android, and iOS platforms. We also recommend signing up for a free Linphone.org SIP account which doesn’t cost you anything. For today, we will be configuring the softphone to register to your new OpenSIPS server.

Once you have downloaded and installed the Linphone client, go into the Preferences menu and make the following changes. Some depend upon your calling platform.

  • Audio Codecs: PCMU, G722, PCMA
  • Video Codecs: VP8, H264
  • Call Encryption: None
  • DTMF: RFC2833 only
  • Send InBand DTMF: OFF
  • Send SIP INFO DTMF: OFF
  • SIP UDP 5060: Enabled
  • SIP TCP 5060: Enabled
  • Allow IPv6: Disabled

Then set up a new SIP Proxy account: Username (7701), Password (as defined), Domain: your FQDN not IP address, Transport: UDP, Outbound Proxy: OFF, Stun Server: stun.linphone.org, ICE: ON, AVPF: OFF, Push Notification: ON, Country Code Prefix: 1 (if required by your commercial SIP provider), Register: YES, Account Enabled: YES. HINT: You can call Alias Names via SIP URI, but you can only register to a SIP account using its actual Username.

Avoiding Lockouts with NeoRouter VPN

By design, Fail2Ban is unforgiving when it comes to failed registrations. A single failed registration will get an IP address banned for a full week. The reason is because the new bad guy strategy is to hit your server once to determine whether anybody is home. Then the creep bombards you later with an endless stream of registration attempts. With our design, nobody will be home when they return. The bad news is a single failed registration attempt by you or your users will also trigger a ban. There are several workarounds. The easiest is to set up the NeoRouter client on each of your machines including your OpenSIPS server and use the 10.0.0.x private network for access. These IP addresses never get banned. Our previous tutorial will walk you through setting up a free NeoRouter server and installing the free NeoRouter clients on your machines. The client software already is installed and running on your OpenSIPS server. It only requires that you log in using nrclientcmd and register to your NeoRouter server to obtain a private IP address. The other option is to install OpenVPN. Our previous tutorial will walk you through that process. The advantage of OpenVPN is that it’s supported directly on many SIP telephone instruments. The 10.8.0.x addresses are already whitelisted by our OpenSIPS installer.

There are other options to unban an IP address which has accidentally been snagged. First, almost all of the cloud providers include a Console option in their web portals. Second, you can log into your server via SSH from any non-blacklisted IP address to remove the banned IP address. Once you’re logged in, simply run this command using the IP address you wish to unban: /root/unban-ip xxx.xxx.xxx.xxx

Choosing Commercial SIP Providers

Recall that you cannot register to a SIP alias on your OpenSIPS server. We’ll take advantage of this restriction in setting up incoming calls from commercial providers’ DIDs. To set up Trunks from commercial providers so that you can not only receive incoming calls but also make outbound calls over their PSTN network connections, you must use providers that support IP address authentication rather than a SIP registration. Many providers support this including our platinum sponsor, Skyetel, as well as providers such as VoIP.ms, Anveo Direct, V1VoIP, and many others. In our OpenSIPS design, you also can use DIDs from providers that support SIP URI forwarding such as CallCentric and LocalPhone; however, you are limited to receiving inbound calls only. VoIP communications really shines here because you don’t have to choose a single provider to meet all of your communications requirements.

Skyetel is by far the easiest provider to set up with OpenSIPS. See our earlier tutorial for a special offer that will get you half-price calling for up to $500. Effective 10/1/2023, $25/month minimum spend required. Once you’re registered on the Skyetel site, add a new EndPoint Group using the IP address of your OpenSIP server and designate UDP 5060 as the access port. Sign up for a DID and map it to the OpenSIPS Endpoint Group. Done. In the OpenSIPS Control Panel, navigate to System -> Dynamic Routing and click Add Gateway. Using the template below, create 5 Proxy gateways for the following Skyetel data centers:

  • skyetel-NW 52.41.52.34
  • skyetel-SW 52.8.201.128
  • skyetel-NE 52.60.138.31
  • skyetel-SE 50.17.48.216
  • skyetel-EU 35.156.192.164

Begin by whitelisting the IP addresses of your SIP providers in /etc/iptables/rules.v4 just below the existing 10.8.0.0/24 rule. The entries should look like this:

-I INPUT -s 52.41.52.34 -j ACCEPT

Once you’ve entered IP addresses for your providers, issue the command: iptables-restart

Next, we need to create what Asterisk users know as an Outbound Route. This tells OpenSIPS to send dialed numbers in 11-digit format to Skyetel for termination. We’ve already created the Dial Plan rule for calling out by dialing 1 plus a 10-digit number. So, while you’re still in the Dynamic Routing section of the OpenSIPS Control Panel, click on the Rules tab at the top of the template. Then click Add Rule. Begin by clicking Add ID button and choosing Group ID 0. In the Prefix field, type 1. Now click the Add GW button 3 times after choosing the Skyetel gateways in the following order from the GW pull-down list: skyetel-nw, skyetel-sw, and skyetel-se. Those are the three currently operational Skyetel gateways. When you’re finished, your template should look like the following. Then click the Add button to save the new rule. Click Reload Server to load the new rule into OpenSIPS. Then repeat this procedure leaving the Prefix field blank so that you can make 10-digit calls as well.

Finally, we need to create what Asterisk users know as an Inbound Route. This tells OpenSIPS where to send incoming calls from our Skyetel DID. OpenSIPS handles inbound routes by defining a User Alias for the Username to which you want to route the incoming DID calls. Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the following template and then click Add.

  • Username: 7701 (the extension to which to route the incoming calls)
  • Domain: opensips.xyz.com (the FQDN of your OpenSIPS server)
  • Alias Username: 18435551212 (the 11-digit Skyetel DID)
  • Alias Domain: 11.12.13.14 (the IP address of your OpenSIPS server)
  • Alias Type: dbaliases

Introducing the VoIP Blacklist

We’ve always dreamed of an effective VoIP Blacklist, and many have tried. But the crowd-sourced VoIP Blacklist at voipbl.org is the real deal. Everybody can post entries (including the bad guys) and, magically, most of the illegitimate entries get sifted out before the next day’s list is released. The list gets populated every night while you sleep. Here are the steps to install the VoIP Blacklist with IPset:

apt update && apt install ipset iptables netfilter-persistent ipset-persistent iptables-persistent
cd /usr/local/sbin
wget http://incrediblepbx.com/voipbl-update
chmod +x voipbl-update
sed -i 's|fail2ban restart|fail2ban restart\n/usr/local/sbin/voipbl-update|' iptables-restart
iptables-restart
ipset list voipbl
ipset list voipbl | wc -l

Then create a cron job in /etc/crontab to run /usr/local/sbin/voipbl-update every day to update the VoIP blacklist.

1 4 * * * root /usr/local/sbin/voipbl-update > /dev/null 2>&1

Congratulations! You now have a functioning OpenSIPS 3 server that can process incoming calls from SIP URIs as well as DIDs. And you can make SIP URI and 11-digit PSTN calls using your SIP softphone that’s registered to your OpenSIPS server. See you next week. Enjoy!

Continue Reading: Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS

Originally published: Monday, October 4, 2021



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.

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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. Nerd Vittles receives referral fees from some VoIP service providers to help cover the costs of our blog. We never recommend particular companies solely to generate commissions. We also test all services that we recommend. []

A New Day: Introducing Incredible PBX 2021 for Ubuntu 20.04

We’ve completed the rollout of Incredible PBX 2021 for Debian 10 and, thanks to the terrific work of @jaminmc, today we’re pleased to introduce Incredible PBX 2021 for Ubuntu 20.04 LTS with its ten-year life cycle.

In addition to @jaminmc, we also want to offer our special thanks to the many talented individuals on the VoIP-Info.org Forum who work through the growing pains of these new releases to bring you open source products at zero cost. Come join the party!

If you’re using on-premise hardware, begin by downloading the ISO image of Ubuntu 20.04 for amd64. Follow our previous tutorials for tips on installation with VirtualBox or VMware ESXi. If you’d prefer to experiment in the cloud for about a penny an hour, open an account at Vultr or Digital Ocean using our referral links that support the Nerd Vittles project. You’ll also get some free credit to try out the service. Then create a new $5/month Ubuntu 20.04, 64-bit instance in your favorite city. Want some cheaper KVM cloud alternatives? Visit the Incredible PBX Wiki.

If your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060-5061 and UDP 10000-20000 traffic to the private IP address of your server. This is required for all of the SIP providers included in the Incredible PBX 2021 build. Otherwise, all inbound calls will fail.

Installing Incredible PBX 2021 on Ubuntu 20.04 Server

Once your Ubuntu 20.04 platform is up and running, login as root using SSH or Putty. If you wish to use the default Asterisk 18 platform, issue the commands below to kick off the install. If you’d prefer to stick with Asterisk 16 for the time being and perhaps upgrade to Asterisk 18 later with the included upgrade script, then edit the script once you download it and change 18 to 16 in line 34 of the script before executing it.

wget http://incrediblepbx.com/IncrediblePBX2021-U.sh
chmod +x IncrediblePBX2021-U.sh
./IncrediblePBX2021-U.sh

At about 5 minutes into the install procedure, you’ll be prompted for your telephone country code. It’s 1 for Canada and the United States. Once the installation completes, reboot and you should be good to go.

Next Steps with Incredible PBX 2021

Before you can manage your PBX through a web browser, you first will need to set the admin passwords for FreePBX, Apache web apps such as Reminders and AsteriDex, and AvantFax (if you are using Incredible Fax). These all can be set by logging into your server as root and issuing the following commands: admin-pw-change, apache-pw-change, and avantfax-pw-change. The AvantFax password can also be reset with a browser by first logging in as admin with a password of password.

Outbound mail functionality needs to be working so that you can receive voicemail messages and faxes by email. To prevent SPAM, most ISPs and ITSPs block messages from downstream mail servers. That would be you. The easiest way to resolve this is to configure SendMail using Gmail as an SMTP Smarthost. You obviously need a Gmail account to implement this and, if you have turned on two-factor authentication for your Gmail account, you also will need to obtain an App password for your Gmail account, and use that in lieu of your regular Gmail password when configuring SendMail. With your Gmail username and password in hand, log into your server as root and run: /root/enable-gmail-smarthost-for-sendmail.

If your Incredible PBX 2021 is hosted with a cloud provider, be advised that many providers do not include a swap file as part of their offering. FreePBX requires a swap file. To add one, issue this command after logging into your server as root: /root/create-swapfile-DO.

To correctly set the time on your PBX, run: /root/timezone-setup.

By default, the voicemail password for each of the configured extensions (701-705) is set to the extension number. This means the user will be prompted to set a voicemail password on the first login to voicemail for each extension. A phone must be registered to the actual extension to access its voicemail account. For example, once a phone is registered to extension 701, the voicemail setup can be accessed by dialing *98701.

If you want to send and receive faxes with Incredible PBX 2021, download the new Incredible Fax 2021 for Ubuntu script, make it executable (chmod +x incred*), and then run the following script while logged into your server as root: /root/incrediblefax2021-ubuntu20.04.sh. When prompted, accept all the defaults. Once the HylaFax and AvantFax components are installed, reboot your server. To send faxes, click on the AvantFax tab in the FreePBX GUI and enter your login credentials (default: admin/password). To receive incoming faxes once you have configured a trunk and DID for your PBX, login to the FreePBX GUI as admin. Navigate to Connectivity -> Inbound Routes. For each of your DIDs on which you wish to receive faxes, select the inbound route and click the Fax tab. Review the Default Inbound Route Fax settings for proper setup.

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 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 a script which will configure settings for dozens of providers in seconds. Once installed, 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. Skyetel is enabled by default and needs no setup on the PBX side.

Configuring Skyetel for Incredible PBX 2021

If you’ve decided to go with Skyetel, here’s the drill. 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 the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

Skyetel does not use SIP registrations to make connections to your PBX. Instead, Skyetel utilizes Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX 2021:

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

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

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

Configuring VoIP.ms for Incredible PBX 2021

To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX 2021 server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls. In the Incredible PBX GUI, be sure to enable the VoIP.ms trunk.

Configuring V1VoIP for Incredible PBX 2021

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

Configuring Anveo Direct for Incredible PBX 2021

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

Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Enable the Trunk. Then click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup. In the Incredible PBX GUI, be sure to enable all of the remaining Anveo trunks.

By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:

cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"

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 (UDP 5061). 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 a Desktop Softphone for Incredible PBX

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

We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

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

DEMO - Apps Demo
123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
TODAY - Today in History
LENNY - The Telemarketer's Worst Nightmare

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store. For Android users, check out the terrific new VitalPBX Communicator. Works flawlessly with Incredible PBX.

Configuring a Softphone Extension on a Smartphone

Adding an Incredible PBX extension to your smartphone gets a little trickier. Whether you’re an iPhone or Android lover, all smartphones use batteries, and you don’t want to drain your battery by running a softphone as a foreground app all the time. Fortunately, you now have some choices in softphones engineered to work without draining your battery. While they all cost money, it’s not much money. We’ve written about all the choices, and you’ll find the links in our Softphone Provider Recommendations on the new Incredible PBX Wiki.

With PJsip extensions, you’re not limited to a single phone connection at a time, and we’ve preconfigured extension 701 to support five simultaneous connections. The setup on the softphone side is simple. For the server, enter the actual IP address of your PBX in the following format: 22.33.44.55:5061. Then enter 701 for the username and enter the password assigned to the 701 extension on your PBX. When an incoming call arrives, all the phones registered to extension 701 will ring simultaneously. Simply answer the call on the phone that is most convenient.

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. 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.

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.

Audio Issues with Incredible PBX 2021

If you experience one-way or no audio on some calls, add your external IP address and LAN subnet in the GUI by navigating to Settings -> Asterisk SIP Settings. In the NAT Settings section, click Detect Network Settings. Click Submit and Apply Settings to save your changes.

Security Considerations with Incredible PBX 2021

Incredible PBX 2021 includes a rock-solid firewall that limits access to preferred providers and individuals whose IP addresses you have whitelisted. Unfortunately, this may not insulate your server from FreePBX 15 irregularities if, in fact, Sangoma’s signing key was compromised in the October 2020 Ransomware Attack. Sangoma either doesn’t know or isn’t telling. Keep in mind that Sangoma didn’t mention the October breach either until someone else exposed it. Sangoma’s latest press release is available here.

The good news is Incredible PBX 2020 and 2021 platforms include a unique ClearlyIP feature that lets us manage which modules and versions can be installed. It works exactly like what ClearlyIP has documented in their must-read blog post, and we’ve built a locked version that rolls back all of the modules to dates before the Sangoma breach. The good news is, with Incredible PBX, you don’t have to jump through all the hoops covered in the ClearlyIP article to fully insulate your server from the Ransomware breach. We’ve done the work for you.

For those with mission-critical platforms, we’d recommend immediate implementation of what follows. For everyone else, it’s your choice whether to wait and see if there is a breach of the signing certificate with malicious modules. If you opt to wait and see, MAKE FREQUENT BACKUPS.

Here’s how to roll back all of your modules to dates before the breach. Login to the FreePBX GUI as admin and navigate to Settings > Advanced Settings. Drop down to the Lock Version field and change 15.19.11.001 to 15.19.11.003. Save your changes and reload your dialplan. Then use Module Admin to roll back any installed modules that are newer than the safe versions shown.

Our extra special thanks goes to Tony Lewis and the ClearlyIP team for providing this invaluable resource. Somehow we knew it would come in handy sooner or later. Unfortunately, that day has come.

Adding Incredible PBX 2021 to an OpenVPN Network

We previously have documented the procedure for creating an OpenVPN server as well as OpenVPN client templates (.ovpn). If you need a refresher, the tutorial is here. To add your Incredible PBX 2021 server to an existing OpenVPN network, begin by creating an incrediblepbx2021.ovpn template on your OpenVPN server. Be sure to comment out or delete the setenv line in the template. Then copy this template to /etc on your Incredible PBX 2021 server. Next, issue the following commands to put the remaining pieces in place:

cp -p /root/openvpn-start /etc/.
echo "[Unit]
Description=openvpn2021
ConditionPathExists=/etc/openvpn-start
After=rclocal.service
[Service]
Type=forking
ExecStart=/etc/openvpn-start /etc/incrediblepbx2021.ovpn
TimeoutSec=0
StandardOutput=tty
RemainAfterExit=yes
PermissionsStartOnly=true
SysVStartPriority=99
[Install]
WantedBy=multi-user.target" > /etc/systemd/system/openvpn2021.service

Finally, enable the new openvpn2021.service and reboot your server. The OpenVPN IP address should now appear on the LAN line in pbxstatus:

systemctl enable openvpn2021.service
reboot

Incredible PBX 2021 Administration

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

add-fqdn is used to whitelist a fully-qualified domain name in the firewall. Because Incredible PBX 2021 blocks all traffic from IP addresses that are not whitelisted, this is what you use to authorize an external user for your PBX. The advantage of an FQDN is that you can use a dynamic DNS service to automatically update the IP address associated with an FQDN so that you never lose connectivity.

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

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

admin-pw-change is used to set the admin password for access to the FreePBX/Incredible PBX web GUI using a browser pointed to the local IP address of your server.

apache-pw-change is used to set the admin password for access to Apache/Incredible PBX apps including AsteriDex and Reminders. This provides a password layer of protection for access to these applications.

avantfax-email-change is used to change the destination email address for incoming faxes.

avantfax-pw-change is used to change your admin password for the AvantFax web console.

iaxmodem-restart is used to restart the modems used to send and deliver faxes. The pbxstatus display will tell you whether the IAXmodems are down.

incrediblebackup2021 makes a backup of critical components on your PBX to a tarball saved in /backup. This should be copied to safe location off-site for a rainy day.

incrediblerestore2021 restores a backup file which has been copied to the /backup folder.

ipchecker is a script which deciphers the public IP addresses associated with whitelisted FQDNs created with add-fqdn on your server. If any of the addresses have changed, the firewall is restarted after updating the IP addresses. By default, it is executed every 10 minutes by /etc/crontab.

licenses.sh displays the license associated with each of the FreePBX modules on your server.

logos-b-gone removes proprietary artwork from your PBX and is no longer necessary with the included IncrediblePBX FreePBX module.

mime-construct is a command-line utility to send emails with attachments.

neorouter-login is a script to add your PBX to a NeoRouter VPN. Tutorial here.

odbc-gen.sh is a script that was run to generate the ODBC settings for Asterisk. Do NOT use it.

openvpn-start is a script to add your PBX to an existing OpenVPN network using an .ovpn config file. Tutorial here.

pbxstatus displays status of all major components of Incredible PBX 2021.

pptp-install is a script to create a PPTP network connection for your PBX. Tutorial here.

purge-cdr-cel-records removes all CDR and CEL records from the MySQL database.

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

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

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

show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX 2021 actually is. A similar listing is available in the GUI at Admin -> Feature Codes.

show-passwords is a script that displays most of the passwords associated with Incredible PBX 2021. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.

sig-fix disables Module Signature Checking in the FreePBX GUI. This should not be necessary unless you have added or edited FreePBX Modules with missing module signatures.

sms-skyetel is a script to send SMS messages using a Skyetel trunk.

sms-voip.ms is a script to send SMS messages using a VoIP.ms trunk.

sms-blast, sms-blaster, and sms-dictator are scripts for message blasting. Tutorial here.

switch-to-php5.6 is a script to disable PHP 7.3 and set PHP 5.6 as the default version for your PBX. PHP 5.6 is required to use AvantFax. It is the default configuration for Incredible PBX 2021. The current default PHP version is displayed in the Apache listing of pbxstatus.

switch-to-php7.3 is a script to disable PHP 5.6 and set PHP 7.3 as the default version for your PBX. You cannot use AvantFax when PHP 7.3 is the default.

timezone-setup is a script to set the timezone for your PBX.

update-IncrediblePBX is a script that runs the Automatic Update Utility whenever you login to your server as root. These updates typically resolve bugs and security issues with your PBX. Do NOT remove it.

upgrade-asterisk16 is a script that runs on Asterisk 16 platforms to upgrade your PBX to the latest release of Asterisk 16.

upgrade-asterisk18 is a script that runs on both Asterisk 16 and 18 platforms to upgrade your PBX to the latest release of Asterisk 18.

wolfram is a script to deploy Wolfram Alpha on your PBX. Tutorial here.

Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"

To remove call forwarding: asterisk -rx "database del CF 705"

Keeping FreePBX 15 Modules Current

We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

rm -f /tmp/*
fwconsole ma upgradeall
fwconsole reload
/root/sig-fix
systemctl restart apache2
/root/sig-fix

 

Originally published: Tuesday, January 26, 2021



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.
 



5 Minutes to Paradise: Incredible PBX in the Cloud for $25/Yr.



We’ve been building turnkey Incredible PBX® servers for virtual machine platforms for many years. Because the servers are built from images, typical install times have been 5 minutes or less with Proxmox, VirtualBox, and VMware ESXi. But the missing piece has always been a similar image install on a true cloud-based platform. This limitation was primarily due to the fact that we don’t own or control the available cloud platforms which typically limit image installs to operating systems such as CentOS, Debian, Ubuntu, and Windows.

Today we have more very good news. Our favorite low-cost hosting provider, Crown Cloud,1 has agreed to offer 5-minute turnkey installs of Incredible PBX 2020 for CentOS and Incredible PBX 2021 for Debian at a jaw-dropping $25 a year. You’ll have a fully-functioning, production-ready KVM platform including a free snapshot with 1GB RAM, 20GB SSD storage, and 1TB of monthly bandwidth in your choice of server locations including Los Angeles and Atlanta in the United States as well as Germany and the Netherlands in Europe. And, unlike all of the other Asterisk® aggregations, Incredible PBX still provides a source code-based platform that can be tailored to meet any special requirements you may need.

We don’t make a nickel on this deal so consider this our special thanks to all of our loyal fans.

Here are the links to sign up for the service and take advantage of these Incredible PBX deals:

UPDATE: 2GB/40GB KVM plans have now been released.

When you sign up for the service, choose any operating system as your default install. Then open a support ticket and request either the Incredible PBX 2020 for CentOS image or the Incredible PBX 2021 for Debian image be added into your template list of available operating systems. Once your server comes on line, open your CrownPanel and choose Reinstall and select Incredible PBX from either the available CentOS or Debian options. In less than 5 minutes, your server will be ready for you to login. Be sure to use SSH and NOT the VNC utility included in CrownPanel. This will assure that your desktop machine’s IP address gets whitelisted in the Incredible PBX firewall. Once you login, the Incredible PBX configurator will prompt you to set passwords for root login, admin login to FreePBX, and admin login credentials for Apache to access AsteriDex and Reminders. After a reboot to initialize your SSH keys, you’ll be ready to go. Add a trunk provider and a softphone, or Incredible PBX SIP phone, or GTI cell phone, and your PBX is fully operational. Check out the CrownCloud Wiki.


Planning Ahead for That Rainy Day

One of our favorite features of Crown Cloud is the free snapshot (a.k.a. backup) at no additional charge. We recommend you take snapshots regularly as you make major changes in your server’s configuration. In this way, if something comes unglued, you can easily restore the snapshot and never miss a beat. You’ll find the Remote Snapshot option in your CrownPanel menu.

Configuring Skyetel for Incredible PBX

If you’ve decided to go with Skyetel, here’s the drill. 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 the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.

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

  • Name: MyPBX
  • Priority: 1
  • IP Address: PBX-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: my.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 ClearlyIP SIP Trunking

For the tightest integration with FreePBX, no SIP provider holds a candle to Incredible PBX SIP Trunking with ClearlyIP. The reason is fairly obvious. The ClearlyIP folks were the original developers of FreePBX. In addition to all of the traditional SIP trunking services, you also get CNAM support and state-of-the-art E911 service which can be deployed in full compliance with Kari’s Law and the Ray Baum Act. If you’re a system integrator and don’t know about your financial liability for failure to comply with the new rules, it’s time to do some reading.

Configuring VoIP.ms for Incredible PBX

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

Configuring V1VoIP for Incredible PBX

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

Configuring Anveo Direct for Incredible PBX

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

Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Enable the Trunk. Then click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup. In the Incredible PBX GUI, be sure to enable all of the remaining Anveo trunks.

By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:

cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"

Configuring a SIP Phone for Incredible PBX

We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which is a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. We obviously recommend the Incredible PBX IP phones from ClearlyIP which are the most versatile.

If you’ve been keeping up with recent Nerd Vittles developments, then you already know that we have just introduced a new Cellular Phone which connects directly to your PBX and serves as a perfect remote extension and traveling companion. You can read all about it here.

Software devices such as the YateClient softphone are another option for desktop machines. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the VoIP-Info.org Forum when you’re ready to get serious about VoIP telephony.

We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.

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

DEMO - Apps Demo
123 - Reminders
947 - Weather by ZIP Code
951 - Yahoo News
TODAY - Today in History
LENNY - The Telemarketer's Worst Nightmare

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store. For Android users, check out the terrific new VitalPBX Communicator. Works flawlessly with Incredible PBX.

For smartphone solutions, visit the Incredible PBX Wiki for our softphone recommendations.

Configuring SendMail with Incredible PBX

In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. We strongly recommend configuring SendMail using either your ISP or Gmail as an SMTP Relay Host. NOTE: If you are using a Gmail account with 2-step verification enabled, you MUST use a Gmail App Key instead of your Gmail account password. You also must enable Less Secure Apps access to the Gmail account.

Configuring a Gmail account with Incredible PBX 2021 is as simple as entering your Gmail credentials. Just run this script: /root/enable-gmail-smarthost-for-sendmail.

Here are the steps using a Gmail account with Incredible PBX 2020:

cd /etc/mail
yum -y install sendmail-cf
hostname -f > genericsdomain
touch genericstable
cd /usr/bin
rm -f makemap
ln -s ../sbin/makemap.sendmail makemap
cd /etc/mail
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
# Stop here and edit client-info (nano -w client-info) in all three lines.
# Replace  user_id with your gMail account name without @gmail.com
# Replace password with your real gMail password OR
#  use your Gmail App Key if 2-step verification is enabled
# Be sure to replace the double-quotes shown above if they don't appear in the file!!!
# Save your changes (Ctrl-X, Y, then Enter)
chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
make
systemctl restart sendmail

Even though these servers are hosted in the cloud, we still recommend using a SmartHost to minimize email delivery problems.

Test outbound mail using this command with your actual email address:

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

On some implementations, you may notice in the FreePBX GUI that the mail queue has failed. Here's the fix:

chmod 777 /var/spool/mqueue
service sendmail restart

Once you are sure your emails are being delivered reliably, here's a sample GUI voicemail configuration for an extension:



Incredible PBX Administration

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

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

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

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

admin-pw-change is used to set the admin password for access to the FreePBX/Incredible PBX web GUI using a browser pointed to the local IP address of your server.

apache-pw-change is used to set the admin password for access to Apache/Incredible PBX apps including AsteriDex and Reminders. This provides a password layer of protection for access to these applications.

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

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

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

show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX actually is. A similar listing is available in the GUI at Admin -> Feature Codes.

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

update-asterisk16 is a script that updates your server to the very latest release of Asterisk 16. You can run it periodically to keep your server up to date with the latest and greatest features.

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

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

Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"

To remove call forwarding: asterisk -rx "database del CF 705"

Getting Started with Incredible Fax

Believe it or not, there still are lots of folks that use faxes in their everyday lives. If you're one of them, Incredible PBX has your back. With Incredible PBX 2021, Incredible Fax is already installed. Simply set your email address for delivery of your faxes in PDF format, and you're all set: /root/avantfax-email-change. With Incredible PBX 2020, begin by logging into your server as root and running incrediblefax2020.sh to install HylaFax and AvantFax on your server. You'll be prompted a dozen or more times for information. Answer no to the secure fax question. For the rest of the prompts, just press ENTER to accept the default entries. Rebooting your server is required when the install finishes.

Once your server is back on line, there will be a new AvantFax tab in the GUI. The default credentials are admin:password. After you enter the username and password, you will be prompted to change your admin password. The old password is still password. Then enter your desired password twice and save the setting. The AvantFax dashboard then will display. If nothing has come unglued, you should see four green Idle icons:



You can Send Faxes from within AvantFax by choosing the Send Fax tab, or you can use one of many HylaFax clients. Google is your friend.

Keeping FreePBX 15 Modules Current

We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:

rm -f /tmp/*
fwconsole ma upgradeall
fwconsole reload
/root/sig-fix
systemctl restart apache2
/root/sig-fix

Where To Go From Here

For a limited time, check out Lenny on our Atlanta demo server by dialing 404-999-0010.

Complete documentation on the ClearlyIP Devices Module is available here.

Complete documentation on the FreePBX GPL Modules is available here.

Complete documentation on the Incredible PBX additions is available here.

An introduction to configuring extensions, trunks, and routes is available here.

Free voicemail transcription with email delivery. Tutorial available here.

Setting Up a VPN for Your PBX: OpenVPN or NeoRouter
 

Originally published: Monday, July 6, 2020  Updated: Saturday, January 16, 2021



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. It has been reported that Norton AntiVirus currently gives a security alert on the Crown Cloud web site. Don't be alarmed. It's a false positive. []

Groundwire for Android & iOS: The Best $10 You’ll Ever Spend



Now we’re getting down to the tough choices in VoIP softphones. You certainly need the Linphone app to make free SIP URI calls worldwide from your mobile phone. But you also need a softphone that connects to Incredible PBX® so that you never miss a traditional call to your home or office PBX. If you want tight integration with Asterisk® and FreePBX®, Clearly Anywhere is the hands-down winner, and you’ve still got until October 31 to take advantage of the $9.99 per year introductory offer. But some folks just bristle at the thought of annual fees for software licenses. If you fall into that category but still want the incoming call reliability offered by Clearly Anywhere, then the one-time $9.99 payment for Acrobits Groundwire on either the Android or iOS platform is your baby.1 Here’s the setup process.

Configuring Incredible PBX 2020 PUBLIC for Acrobits

Before you deploy Acrobits Groundwire on your smartphones, you first must set things up on the PBX side. Here’s how the process works. First, you will need to deploy the PUBLIC version of Incredible PBX 2020 on the Internet. It requires assignment of a fully-qualified domain name (FQDN) for server access. We’ll also need to reconfigure the IPtables firewall and Fail2Ban to support Acrobits Groundwire.

Begin by choosing a cloud provider to host your public server. Here are some of our favorites starting at only a couple bucks a month with an annual subscription. CrownCloud is highly recommended. You’ll need a KVM CentOS 7 platform with at least 1GB of RAM and 10GB of storage. Once your server is operational, log in as root and follow our Incredible PBX 2020 tutorial to begin. When the install finishes, reboot your server and log back in to get the latest updates with the Automatic Update Utility.

Now you’re ready to convert your PBX into a PUBLIC-facing server. You’ll need a fully-qualified domain name for the server. If you don’t have your own domain, you can always obtain a free FQDN from a service such as NoIP.com. With your FQDN in hand, switch over to the Incredible PBX 2020 PUBLIC tutorial to complete the setup. It only takes a few minutes.

Next, we need to whitelist the Acrobits server IP addresses in the IPtables firewall. Begin by editing iptables-custom in the /usr/local/sbin folder. Just above the "# End of Trusted Provider Section" marker, add the following block of code. Then save the file and restart IPtables with the following command: iptables-restart

/usr/sbin/iptables -A INPUT -s 159.65.167.207 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.186.176 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.251.173 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.252.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.253.49 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.252.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.65.253.49 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 159.89.179.103 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 162.243.226.164 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.65.164 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.115.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.182.9 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.184.188 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.190.186 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.210.221 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 165.227.223.68 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.48.91 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.119.203 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -A INPUT -s 167.99.119.244 -p udp -m udp --dport 5060:5069 -j ACCEPT
/usr/sbin/iptables -I INPUT -s 165.227.103.7 -p tcp -m tcp --dport 443 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 7343 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 4998 -j ACCEPT
/usr/sbin/iptables -I INPUT -p tcp -m tcp --dport 24998 -j ACCEPT
/usr/sbin/iptables -I INPUT -p udp -m udp --dport 4998 -j ACCEPT

Finally, we need to whitelist the Acrobits server IP addresses in Fail2Ban to be sure your server doesn’t block any of the Acrobits servers attempting to contact your PBX. Ask us how we know.2 Edit /etc/fail2ban/jail.conf and scroll down to line 34 which begins with ignoreip. Add a space at the end of the existing line and then add the following string of IP addresses without a line break. Save the file and restart Fail2Ban: service fail2ban restart

159.65.167.207 159.65.186.176 159.65.251.173 159.65.252.186 159.65.253.49 159.89.179.103 162.243.226.164 165.227.65.164 165.227.115.186 165.227.182.9 165.227.184.188 165.227.190.186 165.227.210.221 165.227.223.68 167.99.48.91 167.99.119.203 167.99.119.244

Configuring FreePBX for Acrobits Groundwire Access

For every Acrobits Groundwire user, you’ll need to have a PJsip extension to which to register their softphone(s). Keep in mind that multiple softphones of a single user or multiple users can register to the same extension. The first phone to answer an incoming call gets connected to the calling party. Using a web browser, access the FreePBX GUI using your admin credentials. Navigate to Applications -> Extensions -> Add PJsip Extension. Choose a Name and Number for the Extension. Click on the Advanced tab, increase the Max Contacts entry by two for each Acrobits Groundwire softphone that will be connecting to this PJsip extension. Save your PJsip extension setup by clicking Submit. Then Reload the Dialplan.

Acrobits Groundwire Setup on Smartphones

The setup process on the smartphone side is simple. Begin by purchasing the Acrobits Groundwire app from either the App Store or Google Play for your device. Once the client softphone is installed, run the app. You will be prompted for the Username (extension), Password, and Domain:Port of your PBX. If your server is sitting behind a router and you’ve elected to use OpenVPN for your connection, then you’ll need to add the actual Public IP address of your server as the Proxy in Advanced Settings. Don’t forget the :5061 suffix if you’re using a PJsip extension on your PBX.

Once connected, click on the Settings icon in Acrobits Groundwire. Verify that Push Notifications are enabled for Incoming Calls. In setting up your connection to a PJsip extension on your PUBLIC server, don’t forget to tack :5061 onto the server’s FQDN. In the Preferences tab, set a Ring Tone for incoming calls and decide whether to record calls. The Network preference by default will choose WiFi and then Cellular with automatic roll-over when needed. If you want 911 calls routed through your cell carrier instead of through your PBX, choose Number Rewriting. Then click +. For Conditions, choose Equals 911. For Actions, choose Override Dial Action and select gsmCall. Click Done.

We hope you enjoy using the Acrobits Groundwire softphone with Incredible PBX 2020.
 

Originally published: Monday, October 26, 2020   Updated: February 6, 2023



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. As with Vitelity, Acrobits is an Inteliquent company which provides financial support for Nerd Vittles and our Incredible PBX open source project. []
  2. TRUE STORY: We couldn’t get Groundwire to answer incoming calls and finally opened a ticket withe Acrobits. Within 15 minutes, we got a response from Gabriel Baker, who requested a log which was easily generated from the app itself. About 15 minutes after we sent Gabriel the log, he responded noting that our server was blocking the Acrobits IP address which he provided. Issuing the iptables -nL command on our end quickly revealed that Fail2Ban had blacklisted the address. Whitelisting the Acrobits IP addresses in Fail2Ban instantly solved the problem. One would have to conclude that Acrobits has some of the best tech support in the industry based upon our first-hand experience. []

Linphone Rocks: Free SIP Calling to Anybody, Anywhere

We’re rounding out our softphone series today with the perfect complement to Clearly Anywhere for Asterisk® and FreePBX®. Let’s face it. SIP URI dialing from extensions is not Asterisk’s strong suit which is disappointing since SIP URI calling is free. If you’ve been following Incredible PBX® 2020 developments this year, you already know that we’ve pushed everyone to move their VoIP servers to the cloud. You not only get a more flexible PUBLIC platform that lets you go virtually anywhere while preserving home base connectivity for a few bucks a month. A public Asterisk server also lets anyone in the world call your PBX via a SIP URI, and the calls are free from any SIP phone. HINT: Linphone softphones are free!

Today we want to take a fresh look at one of our favorites, Linphone. We call it the perfect complement to Clearly Anywhere because it installs on almost every platform and provides SIP URI communications without any dependence upon a PBX at all. And, just like Clearly Anywhere, multiple devices can connect to the same Linphone account to make and receive calls. Linphone offers free SIP clients for Windows, Mac, Linux, iOS, Android, and many browsers. There’s even GPL source code if that’s your thing. Here are the download links:

To get started, sign up for a free Linphone account. Now simply download the app of your choice from the links above and run it. Then choose the Assistant option and plug in your username and password. Be sure to Allow Push Notifications so that your phone will ring whenever you get an incoming call. Now all you need is a cellular data plan or a wired or wireless network connection. Anyone can reach you at yourname@sip.linphone.org. When friends sign up, their calls also will be free. You can talk as often and as long as you like.

Should you ever lose or forget your password. here’s a link that will let you reset it.

The Linphone feature set is downright impressive. Here’s what you and your friends will be using at zero cost:

CONFIGURATION TIPS: Missing audio or one-way audio is a common problem on SIP calls especially if your phone is sitting behind a NAT-based router. For best results, configure your account in the Linphone app to use UDP for the Transport, Disable the Outbound Proxy, configure stun.linphone.org as the Stun Server, and enable ICE. In Network settings, turn off IPv6 and Media Encryption. In Audio Settings, enable Opus, G.722, PCMU, and PCMA only. In Video Settings, enable both VP8 and H.264. Then close the app and reopen it.

Once you have your Linphone credentials, another option in addition to using one of the SIP clients above is to acquire a stand-alone SIP telephone which can easily be connected to your Linphone SIP account. While there are literally hundreds of SIP telephones from which to choose, here’s a $40 offering from Grandstream that we like. It’s available from Amazon.1 Also consider Incredible PBX phones which take SIP communications to a whole new level.

Unlike other proprietary communications apps, the beauty of using Linphone with its native SIP URI support is you can call any SIP phone in the world for free whether the recipient uses Linphone or not. For example, to annoy your friends and spammers, you can transfer their calls to Lenny: 2233435945@sip2sip.info. And here are some other SIP URI calls you might want to try. Store any or all of them in your Linphone Phonebook.

Your Linphone Friend - friend-name@sip.linphone.org
Yahoo News Headlines - news@demo.nerdvittles.com
Yahoo News Headlines - 951@demo.nerdvittles.com
Weather by Zip Code  - weather@demo.nerdvittles.com
Weather by Zip Code  - 947@demo.nerdvittles.com
Directory Assistance - information@demo.nerdvittles.com
Directory Assistance - 411@demo.nerdvittles.com
Call Any TollFree #  - **1800XXXXXXX@tollfree.future-nine.com

There are now more than 2,000 VoIP networks worldwide that support SIP URI access. Any person or organization with an account on any of these networks can be reached at no cost via SIP URI or via several hundred PSTN numbers. Using a SIP URI dialing prefix, you can call any referenced network@sipbroker.com. For example, *656news@sipbroker.com would reach the Nerd Vittles News Headlines from Yahoo. Or, from a PSTN phone, choose a local access number from the SipBroker worldwide directory, e.g. 702-789-0530 and then dial *656951 at the prompt for Yahoo News from Nerd Vittles. Call enabled PBX extensions the same way.

If you want to associate a phone number with your Linphone SIP URI, you can do it in a couple of ways. First, using a smartphone, you can link your cell number to Linphone within the Linphone app itself. If you have a free DID from IPComms, you can point it to your Linphone SIP URI. If you have a $1/month CallCentric DID, it can also be pointed to your Linphone SIP URI.

There’s one more nice surprise. Linphone accounts work much like the old key telephones and Google Voice setup that we all knew and loved. What that means is you can register the same Linphone account on multiple phones in different places.

Calling Linphone Users from Incredible PBX

Since Linphone doesn’t technically assign phone numbers to users’ accounts, you still need a way for PBX extensions to contact Linphone users. The easiest way is to create a Custom Extension for every Linphone user account you need to reach. In the FreePBX GUI, navigate to Applications -> Extensions -> Add Extension -> Custom Extension. Assign an Extension Number and Name to the custom extension. Then click on the Advanced tab. In the Dial field, enter the SIP URI for the Linphone user, e.g. SIP/username@sip.linphone.org. Click Submit and then Reload the Dialplan.

Originally published: Monday, October 19, 2020



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. This phone requires a wired network connection. 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. []