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The 5-Minute PBX in the Cloud Platform for $2 a Month

If $2.10 a month for a state-of-the-art VoIP PBX isn’t too rich for your blood, then today’s your lucky day. What that buys you is a turnkey, cloud-based PBX platform that is fully installed and configured IN LESS THAN 5 MINUTES! Did we mention the feature set? The Incredible PBX 2027-U image at CrownCloud has an unequaled feature set: Ubuntu 22.04 LTS, Asterisk® 20, and FreePBX® 16 with preconfigured extensions and trunks for dozens of VoIP providers plus voicemail with transcription and email delivery, IVRs, DISA, SMS messaging, conferencing, call queues, paging, intercom, and lots of text-to-speech features to whet your developer appetite: news headlines, weather reports by zip code, today in history, and sample ODBC apps for customer and accounting lookups using MySQL/MariaDB databases. The web server offerings include the latest FreePBX GUI with all of its GPL modules plus WebMin, AsteriDex, and Telephone Reminders. It’s taken us more than a decade to develop this free platform. You get to take advantage of it in less than 5 minutes.

So what’s the difference in today’s $2 a month offering from CrownCloud and the previous offering from RackNerd for $1 a month. The major difference is Incredible PBX 2027-U is an actual image offering at CrownCloud while, at RackNerd, you had to first install the operating system and then go through the entire setup procedure for the Incredible PBX platform of your choice. The CrownCloud scenario takes five minutes while the RackNerd procedure consumes more than an hour assuming all of the repositories upon which Incredible PBX relies are on line and functioning as advertised. What you’ll quickly learn with open source software is this. We rely upon the work product of over 1,000 different individuals and companies, any one of whom can seriously break things either because their software is off-line or because they’ve "improved" something in a way that quickly cascades into a disaster for those of us that relied upon their old way of doing things. If patience and time are your strong suits, then the $1 a month alternative may be a perfect fit. Otherwise, our new CrownCloud image of Incredible PBX 2027-U is a godsend. So let’s get started.

Getting Started at CrownCloud

These 5-minute turnkey installs of Incredible PBX 2027-U for Ubuntu 22.04 are 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, Miami, 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.

It’s worth mentioning that we don’t make a nickel on these cloud 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 the Incredible PBX 2027 deal:

After you sign up for the service, open the CrownPanel application and choose Incredible PBX 2027-U under Application Images for your base install. Take a 5-minute coffee break while your server image is installed. Once your server comes on line, login as root from a desktop machine using the password that was provided. 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 ready to go.

Planning Ahead for That Rainy Day

One of our favorite features of Crown Cloud is the free snapshot (a.k.a. backup) that’s provided 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.

Next Steps with Incredible PBX 2027

Before you can manage your PBX through a web browser, you first will need to set the root password for Ubuntu as well as the admin passwords for FreePBX and Apache web apps such as Reminders and AsteriDex. These all can be set by logging into your server as root and issuing the following commands: passwd, admin-pw-change, and apache-pw-change.

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 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 App password in hand, log into your server as root and run: /root/enable-gmail-smarthost-for-sendmail.

If your Incredible PBX 2027 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 see if you have one, issue the command: free -h. 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.

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 2027

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 2027:

  • 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 2027

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 2027 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 2027

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 2027

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 a 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.


The easiest way to get started is to download a free softphone onto your desktop. We recommend Zoiper 5 for personal use. You can download and install it from here. Once installed, you’ll first need to decipher your extension password for an extension you wish to use to connect to Zoiper. Log into your Linux CLI and switch to the root user as documented above. Then run: /root/show-passwords. Specify 701 and 705 as the starting and ending extensions. Make note of the 701 extension password. Run pbxstatus and make note of your LAN IP address, e.g. 179.xx.yy.zz. Next, start up Zoiper from your desktop and choose the free version. For the desired account enter: 701@179.xx.yy.zz:5061 where 179.xx.yy.zz is the local IP address of your virtual machine. For the password, enter the 701 extension password you deciphered above. Press ENTER twice to complete the connection. When the dialer appears, try out some of the free Incredible PBX applications below.

NOTE: You must use the Keypad option shown in the right window of Zoiper after your call is connected for any app that prompts for keyboard input.

Here are some numbers to try:

123 - Reminders
222 - Timeclock for Employees (try 12345)
223 - AsteriDex Lookup & Dialer (try 335 for Delta Airlines)
947 - Weather by ZIP Code (requires keyboard entry of 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 ten 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. For extension 702, you can change the number of simultaneous connections by clicking the Advanced tab and setting the number in Max Contacts.

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 2027

While it is always a good idea anyway, if you experience one-way or no audio on some calls, be sure to 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.

Adding Fax Support to Incredible PBX 2027

We’ve had to live without faxing on the new Incredible PBX 2027 platforms since the beginning of the year. But those days are over. If you’d like to add fax support to your virtual machine, head over to our new Fax Tutorial to install the components. It only takes a couple minutes.

Adding Incredible PBX 2027 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 2027 server to an existing OpenVPN network, begin by creating an incrediblepbx2027.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 2027 server. The following commands are already in place:

echo "[Unit]
Description=openvpn2027
ConditionPathExists=/etc/openvpn-start
After=rclocal.service
[Service]
Type=forking
ExecStart=/etc/openvpn-start /etc/incrediblepbx2027.ovpn
TimeoutSec=0
StandardOutput=tty
RemainAfterExit=yes
PermissionsStartOnly=true
SysVStartPriority=99
[Install]
WantedBy=multi-user.target" > /etc/systemd/system/openvpn2027.service

We’ve also enabled this openvpn2027.service which will start when you reboot your server. The OpenVPN IP address should now appear on the LAN line in pbxstatus:

systemctl enable openvpn2027.service
reboot

Deploying ODBC Apps with Incredible PBX 2027

Appreciating that many use a PBX to perform actual tasks for your business, we have included two sample ODBC apps to give you a head start. The code for these ODBC apps is stored in odbc.conf in the /etc/asterisk folder. Dialing 222 from any Asterisk extension calls up the TimeClock application which provides access to an employee database. You can enter 12345 as an employee number to see how it works. Dialing 223 from any Asterisk extension connects to the AsteriDex MySQL database and performs a lookup based upon the 3-digit code corresponding to the first three letters of the company name. For example, entering 335 (D-E-L) will call up the entry for Delta Airlines and connect you to their reservations desk once you have an outbound trunk enabled. You could build something similar to look up real-time inventory information from a MySQL database. These are just examples of the thousands of applications that could be quickly developed using ODBC and the Incredible PBX 2027 platform.

Incredible PBX 2027 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 2027 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.

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 2027.

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 2027 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 2027. 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.

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.

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"

Implementing Call By Name with 411

Once you have an Outbound Trunk and Route configured, deploying Call by Name by dialing 411 is simple. The way it works is to pick up any phone connected to your PBX and dial 411. When prompted for the name of the person or company to call, say the name as you entered it in the AsteriDex directory, e.g. Delta Air Lines. The name will then be looked up to decipher the number of the person or company to call. Then the call will be placed using your default outbound route. To deploy Call By Name, simply follow the setup instructions in this Nerd Vittles tutorial.

Introducing Adminer: The Ultimate MySQL Editor

If you’re as sick of phpMyAdmin as we are, you’ll be happy to know there’s a new kid on the block, Adminer. Better yet, the install procedure is a painless, one-minute exercise. The setup procedure for Incredible PBX 2027 is documented here. Once installed, you can connect to Adminer at http://server-ip-address/adminer. You should be prompted for your Apache admin credentials which were configured when you first installed Incredible PBX. Next, enter your MySQL root credentials and Adminer will display in all its glory. DO NOT OPEN PORT 80 FOR PUBLIC ACCESS, OR YOUR ENTIRE PBX WILL BE AT A HACKER’S MERCY!


Keeping FreePBX 16 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

Help If You Can

Many of you know Tony Lewis and especially his tireless efforts over the years in support of the open source community. Our heartfelt condolences go out to Tony and his extended family upon the death of his brother, Dan, last week. In the midst of renovating their home, Dan leaves behind his wife and four young children to pick up the pieces. If you’re able, they could use your financial help. A Go Fund Me campaign was launched in Dan’s honor here.

Originally published: Tuesday, April 25, 2023



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

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

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

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



The Beauty of Incredible PBX: ClearlyIP Module Management



When Incredible PBX was released, you may recall that we jettisoned reliance on FreePBX® repositories and adopted Clearly IP’s far superior repository setup for FreePBX module management. Most of you have never noticed… until today. As lady luck would have it, Sangoma recently released a buggy version of the User Control Panel (UCP) module which literally blows up when you next attempt to access UCP.

One of the hidden beauties of Incredible PBX is that we now have version control over FreePBX modules thanks to Clearly IP’s Mirror Administration application. What this means is we can restrict which versions of every FreePBX module get installed on every Incredible PBX server. So when a problem such as the UCP bug causes a serious operational issue for our users, we can simply use Mirror Admin to back rev the UCP module to a working release, version 16.0.36 in this case.

The other hidden advantage of Incredible PBX is we maintain dozens of honeypot servers around the world running many different versions of Linux, Asterisk, and FreePBX. So, when this issue popped up, it was a simple matter to visit various sites until we found one using a functional version of the User Control Panel. Then a simple tweak of the default UCP version in Mirror Admin meant administrators could quickly restore UCP functionality. Here’s how.

Begin by logging into the FreePBX GUI with your admin credentials. Navigate to Module Administration and choose the Check Online option. Scroll down to the User Control Panel module and choose Force Download and Downgrade. Click the Process button to migrate back to UCP 16.0.36, and you’re back in business. Enjoy… and tell your friends!

Originally published: Monday, April 17, 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.
 



Happy New Year: It’s Incredible PBX 2027 for Debian 11

These past two weeks to celebrate the New Year, we’ve introduced Incredible PBX 2027 for Rocky 8 with AMD64-compatible hardware and Incredible PBX 2027 for Ubuntu 22.04 with AMD64 and ARM64 hardware support. Today we’re pleased to introduce Incredible PBX 2027 for Debian 11 featuring Asterisk® 20 LTS support until the fall of 2027. This release comes with full support for FreePBX® 16 as well as all the Incredible PBX goodies to which you’ve become accustomed.

UPDATE: Proxmox image of Incredible PBX 2027-D now available in the Incredible PBX Repository.

Again we want to offer our thanks to the many talented individuals on the VoIP-Info.org Forum who have assisted us in working through the growing pains of bringing you these new open source products at zero cost. And our special thanks to @kenn10 for his Herculean efforts refining the Incredible PBX 2027 install scripts. Come celebrate the New Year with us and join the party!

If you’re using on-premise hardware, download Debian 11, codenamed bullseye, netinst, for 64-bit PC (amd64). 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 Debian 11, 64-bit instance in your favorite city. Want some cheaper KVM cloud alternatives? Visit the Incredible PBX Wiki for tips.

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 2027 builds. Otherwise, all inbound calls will fail.

Installing Incredible PBX 2027 on Debian 11 Server

If you’re building the Debian platform from the ISO, boot your server from the ISO, choose your Language, choose your Location, and choose your Keyboard. Wait for the network to be configured. Then leave the Hostname set as debian. Leave the Domain blank. Set your root password. Create a New User and Password. Set the TimeZone for your clock. Accept the defaults to partition disks. Write the changes to disk. Wait for the base system to be installed. Answer NO at the Scan More Media prompt. Select a Mirror for Package Manager that is close to you. Leave the proxy information blank. At the Software Selection prompt, choose only the bottom two options to get the basics and SSH. Load the GRUB boot manager and select your drive. Then reboot. After reboot, login as the new user and issue these commands:

# set root password
su root
passwd
exit
exit
# log back in as root
# delete temporary user
userdel -r temp-user-name
# decipher server IP address
ip a
# try logging in via SSH: ssh root@ip-address

If you cannot login via SSH or Putty as root then, from the console while logged in as root, issue these commands:

apt install nano -y
cd /etc/ssh
nano -w sshd_config
# change the following entries as shown here:
  PermitRootLogin yes
  PasswordAuthentication yes
# save the file and restart SSH
# Ctrl-X, Y, and press ENTER key
systemctl restart ssh

Once your Debian 11 platform is properly configured, login as root using SSH or Putty. Issue the commands below to kick off the install:

apt install wget tar nano -y
wget http://incrediblepbx.com/IncrediblePBX2027-D.sh
chmod +x IncrediblePBX2027-D.sh
./IncrediblePBX2027-D.sh

Setting a Hostname for Incredible PBX 2027

Once your server is up and running, you’ll need to set a hostname for the machine that is resolvable on the Internet. Failure to do so will make access to the FreePBX GUI a painfully delayed process. If you don’t have an FQDN that can be used, you can use the default: noreply.incrediblepbx.com. To change it, edit /etc/hosts and /etc/hostname. Also enter your new FQDN with the command: hostname myfqdn.com.

Next Steps with Incredible PBX 2027

Before you can manage your PBX through a web browser, you first will need to set the root password for Debian as well as the admin passwords for FreePBX and Apache web apps such as Reminders and AsteriDex. These all can be set by logging into your server as root and issuing the following commands: passwd, admin-pw-change, and apache-pw-change.

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 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 App password in hand, log into your server as root and run: /root/enable-gmail-smarthost-for-sendmail.

If your Incredible PBX 2027 is hosted with a cloud provider, be advised that some providers do not include a swap file as part of their offering. To see if a swapfile is enabled, issue the command: free -h. 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.

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 2027

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 2027:

  • 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 2027

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 2027 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 2027

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 2027

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 ten 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. For extension 702, you can change the number of simultaneous connections by clicking the Advanced tab and setting the number in Max Contacts.

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 2027

While it is always a good idea anyway, if you experience one-way or no audio on some calls, be sure to 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.

Adding Incredible PBX 2027 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 2027 server to an existing OpenVPN network, begin by creating an incrediblepbx2027.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 2027 server. The following commands are already in place:

echo "[Unit]
Description=openvpn2027
ConditionPathExists=/etc/openvpn-start
After=rclocal.service
[Service]
Type=forking
ExecStart=/etc/openvpn-start /etc/incrediblepbx2027.ovpn
TimeoutSec=0
StandardOutput=tty
RemainAfterExit=yes
PermissionsStartOnly=true
SysVStartPriority=99
[Install]
WantedBy=multi-user.target" > /etc/systemd/system/openvpn2027.service

We’ve also enabled this openvpn2027.service which will start when you reboot your server. The OpenVPN IP address should now appear on the LAN line in pbxstatus:

systemctl enable openvpn2027.service
reboot

Incredible PBX 2027 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 2027 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.

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 2027.

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 2027 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 2027. 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.

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.

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"

Implementing Call By Name with 411

Once you have an Outbound Trunk and Route configured, deploying Call by Name by dialing 411 is simple. The way it works is to pick up any phone connected to your PBX and dial 411. When prompted for the name of the person or company to call, say the name as you entered it in the AsteriDex directory, e.g. Delta Air Lines. The name will then be looked up to decipher the number of the person or company to call. Then the call will be placed using your default outbound route. To deploy Call By Name, simply follow the setup instructions in this Nerd Vittles tutorial.

Keeping FreePBX 16 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: Monday, January 16, 2023



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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.
 



Happy New Year: It’s Incredible PBX 2027 for Ubuntu 22.04

Last week to celebrate New Year’s Day, we introduced Incredible PBX 2027 for Rocky 8 with AMD64-compatible hardware. Today we’re pleased to introduce Incredible PBX 2027 for Ubuntu 22.04 LTS with its ten-year life cycle to complement Asterisk® 20 featuring LTS support until the fall of 2027. Both new AMD64 releases come with full support for FreePBX® 16 as well as all the Incredible PBX goodies to which you’ve become accustomed.

We also want to offer our thanks to the many talented individuals on the VoIP-Info.org Forum who have assisted us in working through the growing pains of bringing you these new open source products at zero cost. And our special thanks to @kenn10 for his Herculean efforts refining the Incredible PBX 2027 install scripts. Come join the party!

If you’re using on-premise hardware, begin by downloading the Live Server ISO image of Ubuntu 22.04 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 22.04, 64-bit instance in your favorite city. Want some cheaper KVM cloud alternatives? Visit the Incredible PBX Wiki for tips.

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 2027 build. Otherwise, all inbound calls will fail.

Installing Incredible PBX 2027 on Ubuntu 22.04 Server

If you’re building the Ubuntu platform from the ISO, select the option to install a Minimal Server Platform. Then you’ll need to create a temporary user as part of the install. Enable installation of SSH and no additional components. After reboot, login as the new user and issue these commands:

# set root password
sudo passwd root
exit
# log back in as root
# delete temporary user
userdel -r temp-user-name
# decipher server IP address
ip a
# try logging in via SSH: ssh root@ip-address

On desktop machines, if you experience a 2-minute delay on boot up waiting for network to be configured, issue these two commands after logging in as root:

systemctl disable systemd-networkd-wait-online.service
systemctl mask systemd-networkd-wait-online.service

If you cannot login via SSH or Putty as root then, from the console while logged in as root, issue these commands:

apt install nano -y
cd /etc/ssh
nano -w sshd_config
# change the following entries as shown here:
  PermitRootLogin yes
  PasswordAuthentication yes
# save the file and restart SSH
# Ctrl-X, Y, and press ENTER key
systemctl restart ssh

ALERT: On some cloud platforms, e.g. RackNerds, they reportedly provide a non-interactive version of Ubuntu 22.04. So, after completing the steps above, you will need to run the following script: /usr/local/sbin/unminimize. Next, run: apt update && apt upgrade. Finally, reboot the server and log back in as root to continue.

Once your Ubuntu 22.04 platform is properly configured, login as root using SSH or Putty. Issue the commands below to kick off the install:

apt install wget tar nano -y
wget http://incrediblepbx.com/IncrediblePBX2027-U.sh
chmod +x IncrediblePBX2027-U.sh
./IncrediblePBX2027-U.sh

On homespun, ISO-built platforms, you will be prompted within a minute or so on whether to preserve your existing SSH configuration. Press ENTER to accept the default.

Setting a Hostname for Incredible PBX 2027

Once your server is up and running, you’ll need to set a hostname for the machine that is resolvable on the Internet. Failure to do so will make access to the FreePBX GUI a painfully delayed process. If you don’t have an FQDN that can be used, you can use the default: noreply.incrediblepbx.com. To change it, edit /etc/hosts and /etc/hostname. Also enter your new FQDN with the command: hostname myfqdn.com.

If you’re running your PBX in the Oracle Cloud, this message thread will show you how to set the hostname there.

Next Steps with Incredible PBX 2027

Before you can manage your PBX through a web browser, you first will need to set the root password for Ubuntu as well as the admin passwords for FreePBX and Apache web apps such as Reminders and AsteriDex. These all can be set by logging into your server as root and issuing the following commands: passwd, admin-pw-change, and apache-pw-change.

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 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 App password in hand, log into your server as root and run: /root/enable-gmail-smarthost-for-sendmail.

If your Incredible PBX 2027 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.

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 2027

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 2027:

  • 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 2027

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 2027 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 2027

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 2027

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 ten 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. For extension 702, you can change the number of simultaneous connections by clicking the Advanced tab and setting the number in Max Contacts.

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 2027

While it is always a good idea anyway, if you experience one-way or no audio on some calls, be sure to 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.

Adding Incredible PBX 2027 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 2027 server to an existing OpenVPN network, begin by creating an incrediblepbx2027.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 2027 server. The following commands are already in place:

echo "[Unit]
Description=openvpn2027
ConditionPathExists=/etc/openvpn-start
After=rclocal.service
[Service]
Type=forking
ExecStart=/etc/openvpn-start /etc/incrediblepbx2027.ovpn
TimeoutSec=0
StandardOutput=tty
RemainAfterExit=yes
PermissionsStartOnly=true
SysVStartPriority=99
[Install]
WantedBy=multi-user.target" > /etc/systemd/system/openvpn2027.service

We’ve also enabled this openvpn2027.service which will start when you reboot your server. The OpenVPN IP address should now appear on the LAN line in pbxstatus:

systemctl enable openvpn2027.service
reboot

Incredible PBX 2027 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 2027 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.

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 2027.

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 2027 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 2027. 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.

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.

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"

Implementing Call By Name with 411

Once you have an Outbound Trunk and Route configured, deploying Call by Name by dialing 411 is simple. The way it works is to pick up any phone connected to your PBX and dial 411. When prompted for the name of the person or company to call, say the name as you entered it in the AsteriDex directory, e.g. Delta Air Lines. The name will then be looked up to decipher the number of the person or company to call. Then the call will be placed using your default outbound route. To deploy Call By Name, simply follow the setup instructions in this Nerd Vittles tutorial.

Keeping FreePBX 16 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: Monday, January 9, 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.
 



Happy New Year: Introducing Incredible PBX 2027



If we have one complaint with open source VoIP telephony, it would be this. Commercial PBXs used to last for a decade or more. Now, most open source VoIP platforms are measured in months. So, we have a better idea. We’re going to tell you up front how long Incredible PBX 2027 with Rocky 8 and Asterisk® 20 and FreePBX® 16 is going to be supported. And, for procrastinators, you’ll have until the fall of that year to line up an alternative. Having said all of that, we are pleased to introduce Incredible PBX 2027-R for Rocky 8 and AlmaLinux 8 featuring Asterisk 20 and your choice of FreePBX 15 or 16.

UPDATE: Proxmox image of Incredible PBX 2027-R now available in the Incredible PBX Repository.

CAUTION: Because of frequent changes in Rocky 8 that regularly "break things," we no longer recommend it for production use with Incredible PBX>

Begin by downloading and installing Rocky 8 Minimal. If you prefer AlmaLinux, download and install their minimal image from here. If you prefer a cloud-based platform, consider CrownCloud for $25/year or our Platinum sponsor, Vultr. Both providers host Rocky 8 and AlmaLinux 8 AMD64 images in addition to other operating systems so platform setup is quick.

Once you have your platform up and running, login as root and issue the following commands to kick off the Incredible PBX 2022 install:

cd /root
yum -y install wget tar
wget http://incrediblepbx.com/incrediblepbx2027.tar.gz
tar zxvf incrediblepbx2027.tar.gz
rm -f incrediblepbx2027.tar.gz
./IncrediblePBX2027.sh

Once the Phase I install finishes, your server will reboot. Log back in as root and run the installer a second time:

cd /root
./IncrediblePBX2027.sh

When the install finishes, reboot your server and login again as root. Incredible PBX will run the Automatic Update Utility to bring your server up to current specs.

If you chose to install AlmaLinux 8, issue the following command while still logged into the Linux CLI as root:

sed -i 's|Rocky 8|AlmaLinux 8|' /usr/local/sbin/pbxstatus

Just a few more preliminary steps, and you’ll be ready to go:

1. Create a secure root password for your server by issuing the command: passwd

2. Remove temporary user account: userdel -r temp-user-name

3. Set up an admin password for browser access to the FreePBX GUI: /root/admin-pw-change

4. Set up an admin password for Apache access to AsteriDex and Reminders: /root/apache-pw-change

5. Set the correct time zone for your server: /root/timezone-setup

6. If free -h shows no swapfile enabled, issue this command: /root/create-swapfile-DO

As with all Incredible PBX builds, a preconfigured Linux firewall is included which blocks all access except from whitelisted IP addresses. You can whitelist additional IP addresses using the /root/add-ip script or /root/add-fqdn.

You’re now ready to explore the Incredible PBX feature set using the tutorials available on the Incredible PBX Wiki.

Configuring Gmail Smarthost for Incredible PBX 2027

One piece you’ll need for many tasks in Incredible PBX 2027-R is a way to send emails from your server. The easiest method to accomplish this is to configure SendMail to use a Gmail SmartHost. We’ve included a script in the /root folder to make it painless. Simply run: /root/enable-gmail-smarthost-for-sendmail. When prompted, enter your Gmail account name (without @gmail.com) and then enter a Gmail Application Password, not your standard Gmail password. If you don’t have one, you can obtain one here. Once you’ve configured SendMail, try things out by sending an email to any valid email address:

echo "test" | mail -s testmessage someone@somewhere.vom

Upgrading to FreePBX 16 with Incredible PBX 2027-R

By default, Incredible PBX 2027-R comes preconfigured with the tried-and-true FreePBX 15 GUI. But there’s a new kid on the block, FreePBX 16. If you’d like to take it for a spin, the upgrade is painless. Simply issue the following command while logged into your server as root: /root/upgrade-to-FreePBX16.


Audio Issues with Incredible PBX 2027-R

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. While you’re there, click on the chan/pjsip tab and set Allow Transports Reload to NO.

Choosing SIP Providers for Incredible PBX 2027-R

Incredible PBX 2027-R comes preconfigured with support for five SIP extensions and a dozen major SIP providers including Skyetel, VoIP.ms, V1VoIP, and Anveo Direct. We hope you’ll choose Skyetel not only because they financially support Nerd Vittles and our open source projects, but also because it is a clearly superior platform offering crystal-clear communications and triple-redundancy so you never miss a call. Skyetel also sets itself apart from the other providers in the support department. They actually respond to issues, and there’s never a charge. As the old saying goes, they may not be the cheapest, but you get what you pay for. Even without taking advantage of Nerd Vittles half-price offer on up to $500 of Skyetel services, they’re still dirt cheap compared to the Bell Sisters and cable companies. Skyetel is so sure you’ll love their service that they give you a $10 credit to kick the tires before you ever spend a dime. Traditional DIDs are $1 per month. Outbound conversational calls are $0.012 per minute. Incoming conversational calls are a penny a minute, and CallerID lookups are $0.004. You only pay for minutes you use. Once you’re satisfied with the service and fund your account, you can port in your existing DIDs at no cost for 60 days after signup. In short, you have nothing to lose by trying out the Skyetel service.

Configuring Skyetel for Incredible PBX 2027-R

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 typically 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 2020:

  • 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 2027-R

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 2027-R 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 2027-R

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 2027-R

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 BulkVS for Incredible PBX 2027-R

Unlike traditional telephony, you have nothing to lose by configuring multiple trunks with Incredible PBX. If you don’t make calls using one or more of the trunks, you pay nothing. Another more recent provider with excellent rates is BulkVS. We’ve covered the BulkVS setup in a separate tutorial if you’d like to give them a try.

Configuring a Softphone for Incredible PBX 2027-R

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. 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 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 Headlines
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.

Introducing WebMin for Incredible PBX 2027-R

WebMin is also installed and configured as part of the base install. The root password for access is the same as your Linux root password. We strongly recommend that you not use WebMin to make configuration changes to your server. You may inadvertently damage the operation of your PBX beyond repair. WebMin is an excellent tool to LOOK at how your server is configured. When used for that purpose, we highly recommend WebMin as a way to become familiar with your Linux configuration. To access WebMin, open a browser to http://server-ip-address:9001.

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 as root 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 2020 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 2020 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 2020. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.

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

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

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"

Implementing Call By Name with 411

Once you have an Outbound Trunk and Route configured, deploying Call by Name by dialing 411 is simple. The way it works is to pick up any phone connected to your PBX and dial 411. When prompted for the name of the person or company to call, say the name as you entered it in the AsteriDex directory, e.g. Delta Air Lines. The name will then be looked up to decipher the number of the person or company to call. Then the call will be placed using your default outbound route. To deploy Call By Name, simply follow the setup instructions in this Nerd Vittles tutorial.

Keeping FreePBX 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.

Originally published: Sunday, January 1, 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.
 




 

Tired of Waiting on Hold? Here’s a Simple Asterisk Fix



We’ve all placed calls to support lines that keep you waiting on hold for what seems like an eternity. While some companies now offer a call back service that holds your place in the queue, many still do not. One of our users recently asked whether there was a simple way with Asterisk® to preserve your place in line without tying up your phone waiting for the other party to finally pick up the call. Here’s what we came up with.

The idea here is to use Unattended Transfer to transfer your end of the call to an extension that plays a message to dial zero to be reconnected with the calling party once your turn in the call queue finally arrives. To send your call to Pseudo-Hold, dial ##4653# (4653 spells HOLD). On Incredible PBX platforms, this will initiate an unattended transfer of your call and free up your phone.

Now we only need a little Asterisk dialplan code plus a recorded message telling the other party to press 0 to be immediately reconnected with us when they come on the phone. When the other party presses 0, the dialplan will transfer the call back to the extension that transferred the call to what we are referring to as Pseudo-Hold.

To begin, navigate to the custom folder in either /var/lib/asterisk/sounds/custom or /var/lib/asterisk/sounds/en/custom depending upon your version of Asterisk. Now download and install both the recording and the dialplan snippet:

wget http://incrediblepbx.com/presszero.tar.gz
tar zxvf presszero.tar.gz
rm -f presszero.tar.gz
sed -i '\:// BEGIN Zero:,\:// END Zero:d' /etc/asterisk/extensions_custom.conf
sed -i '/\[from-internal-custom\]/r dialplan.txt' /etc/asterisk/extensions_custom.conf
rm -f dialplan.txt
asterisk -rx "dialplan reload"

For those that are curious, the dialplan.txt snippet looks like this:

;# // BEGIN Zero Reconnect
exten => 4653,1,Answer
exten => 4653,2,NoOp(Dialed Peer Number: ${DIALEDPEERNUMBER})
exten => 4653,3,Background(custom/PressZeroToReconnect)
exten => 4653,4,WaitExten(2)
exten => 0,1,Background(connecting)
exten => 0,2,Dial(SIP/${DIALEDPEERNUMBER})
exten => t,1,Goto(4653,3)
exten => i,1,Goto(4653,3)
exten => h,1,Hangup
;# // END Zero Reconnect

To walk you through how it works, here we go. When you press ##4653#, the call is transferred to the 4653 extension shown above. In line 1, it answers the call. In line 2, it deciphers the extension from which the call was transferred. In line 3, we play the Press Zero to Reconnect message. In step 4, we wait for the other party to press 0. If they don’t within 2 seconds, we loop back up to line 3 and play the message over and over again. If the other party presses zero, we drop down to 0,1 which plays a message saying "Connecting." Then 0,2 sends the call back to the extension that originally transferred the call to 4653.

Originally published: Monday, December 12, 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.
 




 

Adiós Twitter: Introducing Mastodon for the VoIP Community



#FollowFriday seemed an appropriate day to introduce Mastodon, our decentralized solution to the Twitter meltdown. If you’ve been living under a rock for the past couple weeks, here’s a quick snapshot. Elon Musk buys Twitter for $44 billion. Elon fires half of existing Twitter staff. Elon bans Twitter employees from working at home. Elon, the Free Speech champion, fires Twitter employees that criticize his misleading tweeets. Elon gives remaining Twitter employees an ultimatum to sign a pledge to work under grueling working conditions or leave with three months of severance pay. 75% of remaining employees refuse to sign the pledge, and Elon locks the doors.



Meanwhile, the Mastodon alternative explodes to over 7 million users in a single week. Today we want to introduce you to Mastodon and encourage each of you to take it for a test drive. It’s free. It’s decentralized. And it’s incredibly easy to take advantage of the independence that the Mastodon social media platform offers without worrying about a billionaire ever putting the platform in jeopardy. A week ago we couldn’t spell Mastodon, and today we have a thriving, independent Mastodon platform that offers you almost everything that it took many of us years to build on Twitter. You can sign up for a free account here. Then have a look at our Public (a.k.a. Federated) platform here.


The beauty of the Mastodon design is you get the best of both worlds. You can enjoy the content of everyone that anyone on your Mastodon instance chooses to follow while also participating on a local platform that is dedicated to the interests of the Incredible PBX and VoIP community. There is no other VoIP, Asterisk, or FreePBX platform for Mastodon today!


Getting Started. Once you’ve signed up for an account, the first thing you will want to do is to Enable the Advanced Web Interface in Settings. This provides simultaneous views of both your Local community and the Federated community through the web interface. The Federated view already is populated with a feature-rich content collection much like what it took many of us years to create as Twitter users. We should note that Mastodon is a very mature platform offering web access as well as a number of excellent apps for iPhone and Android users. We’ve also started a thread on VoIP-Info.org forum with numerous tips and tricks to get you up to speed on Mastodon. Come join the party!

Here’s our favorite Getting Started with Mastodon article. Start here.

Originally published: Friday, November 18, 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.
 




 

SpeechGen.io: A Near Perfect TTS Offering for Asterisk



Over the years, we’ve covered numerous commercial and free text-to-speech (TTS) offerings for Asterisk® including gTTS, PicoTTS, Amazon’s Polly TTS, IBM TTS, Google TTS, FLITE, and Festival. But SpeechGen’s commercial offering sets it apart both in terms of quality and cost. At a $9.99 price point for 100,000 characters with an easy-to-deploy AGI interface, SpeechGen is almost a perfect fit for Asterisk TTS applications. If you decide to deploy SpeechGen after trying out our demo, we hope you’ll consider using our referral link which helps offset the cost of providing today’s Weather by ZIP Code demo for everyone:

For those who have deployed Incredible PBX, we have a one-minute install script which will put all the pieces in place to use SpeechGen with our News Headlines and Weather by ZIP Code applications. Simply issue the commands below after logging into your server as root. Next, edit speechgen.php and insert your SpeechGen API token and your email address. Then dial 951 or 947 from any extension connected to your PBX to retrieve today’s news headlines or weather forecast for any U.S. zip code.

cd /var/lib/asterisk/agi-bin
wget http://incrediblepbx.com/speechgen.tar.gz
tar zxvf speechgen.tar.gz
rm -f speechgen.tar.gz
./install-speechgen-dialplan.sh
nano -w speechgen.php

Send us your own TTS applications for Asterisk. We’d love to publish them.

Originally published: Monday, October 31, 2022



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