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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiDeal of the Year: OBi2182 Color WiFi Phones now $59.99
One year and counting… If you still have some Google Voice numbers lying around or if you’d like to join the party a little late, now is your second chance. Amazon now is selling the OBi2182 phones to U.S. residents with native Google Voice as well as WiFi, Bluetooth, HD Voice, speakerphone, and a power adapter currently selling for $69.99. Hold, transfer, and conferencing support are included in the phone. Here’s our Amazon referral link to keep the lights on.
We were big fans of OBiHai once upon a time. But then they sold out to Polycom. And they subsequently sold out to Plantronics. And support went from bad to worse along the way. But things appear to have stabilized so this offering may just be one of the best deals on the planet. Not only do you get unlimited free calling throughout the U.S., Canada, and Ukraine but you also get some of the best pricing ever on most international calls. While we no longer have any dealings with ObiHai, Polycom, or Plantronics, we know many of our followers were left high and dry when Google pulled the plug on Google Voice for Asterisk®. Here’s a tip.
Adding one or more of these phones makes a perfect supplement to your Asterisk platform making it easy to add free calling to your home or home office PBX. And it’s not too late to order a Google Voice number if you’re new to the game. Configuring a SIP or PJsip connection as well as Bluetooth connectivity to your cellphone is just as simple as adding Google Voice trunks using the OBiTalk web interface shown below. Trunk setups take less than a minute apiece. The phone supports 12 lines. And WiFi calls are crystal-clear thanks to HD Voice.
Grab an OBi2182 while you still can! An Admin Guide to the OBi2182 is available here.
Originally published: Monday, May 9, 2022
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Interconnecting Asterisk Servers with PJsip and OpenVPN
It’s been several years since we discussed interconnecting Asterisk® servers so today we want to do a version refresh using PJsip Trunking. We also want to show you how easy it is to secure the communications path by setting up the trunks using OpenVPN connections. When we’re finished, you’ll have a FREE way to call between sites using FreePBX® Outbound Routes. Because Incredible PBX comes preconfigured with all the components you’ll need, we’ll use that platform to further simplify the deployment. We’ll be interconnecting two Asterisk servers today, but you can use the same methodology to connect numerous sites.
Deploying OpenVPN with Asterisk Servers
To begin, you’ll want to get all of the sites configured with a virtual private network using OpenVPN. Our tutorial will walk you through the process. Keep in mind that all current releases of Incredible PBX are preconfigured to let you drop in your OpenVPN client credentials by naming them incrediblepbx.ovpn and copying the file into the /etc directory. Rebooting your server will bring up the virtual private network with a 10.8.0.x IP address.
Configuring PJsip Trunks on Your Asterisk Servers
If you remember yesteryear’s knuckle drill configuring SIP or IAX trunks for Asterisk connectivity, you’re in for a pleasant surprise using PJsip trunking with FreePBX. Using the GUI, create a new PJsip trunk for every site to which you want to establish a connection. A similar PJsip trunk must be created on the other site as well. If you’re just interconnecting two servers, then using the City locations for the Trunk Names will suffice. But, if there are more than two servers, specify unique names for each end of every PJsip connection, e.g.
NewYork1 <-> London1 NewYork2 <-> Washington1 NewYork3 <-> Miami1 London2 <-> Washington2 London3 <-> Miami2 Washington3 <-> Miami3
For today, we’ll interconnect a server in New York and London, but we’ll plan for the future and use London1 for the Trunk Name on the New York server and NewYork1 for the Trunk Name on the London server.
Let’s begin by configuring the London1 trunk on the New York server. After entering the London1 Trunk Name in the General tab, click on the pjsip Settings tab. In the General tab, leave the defaults in the first four fields. Then change the Registration field to None. For the SIP Server field, enter the OpenVPN IP address of the London server, e.g. 10.8.0.2. Because we’re using Incredible PBX, the PJsip port is 5061 so enter that in the SIP Server Port field. If you’re using a different flavor of FreePBX, enter the appropriate port number for PJsip on your platform. Next, click on the Advanced tab and enter the London server’s OpenVPN address in the Match (Permit) field, e.g. 10.8.0.2. In the Codecs tab, make note of the enabled codecs and make certain that the entries match on all of your servers. Click the Submit button to save your settings and then reload your dialplan.
Now let’s configure the NewYork1 trunk on the London server. After entering the NewYork1 Trunk Name in the General tab, click on the pjsip Settings tab. In the General tab, leave the defaults in the first four fields. Then change the Registration field to None. For the SIP Server field, enter the OpenVPN IP address of the New York server, e.g. 10.8.0.1. Because we’re using Incredible PBX, the PJsip port is 5061 so enter that in the SIP Server Port field. If you’re using a different flavor of FreePBX, enter the appropriate port number for PJsip on your platform. Next, click on the Advanced tab and enter the New York server’s OpenVPN address in the Match (Permit) field, e.g. 10.8.0.1. In the Codecs tab, make certain that the entries match those on your New York server. Click the Submit button to save your settings and then reload your dialplan. Here’s how it looks in the FreePBX GUI:
Use Outbound Routes to Interconnect Extensions
To keep things simple, let’s assume both your New York and London servers have extensions 701-705. To call an extension on the other server, we will simply dial 9 and then the 3-digit extension, e.g. dialing 9701 on the New York server will ring 701 on the London server and dialing 9701 on the London server will ring 701 on the New York server.
Create an Outbound Route on the New York server called London specifying London1 for the Trunk Sequence in the Route Settings tab. In the Dial Patterns tab, enter 9 in the Prefix field and XXX for the Match Pattern. Click Submit to save your settings and then reload dialplan.
Create an Outbound Route on the London server called New York specifying NewYork1 for the Trunk Sequence in the Route Settings tab. In the Dial Patterns tab, enter 9 in the Prefix field and XXX for the Match Pattern. Click Submit to save your settings and then reload dialplan.
If you’re interconnecting more than two sites, then you probably will want to designate a specific Prefix for every City so that users can travel between sites and use the same methodology to reach the same extensions from every location.
You can test things out using softphones by registering 701 to an extension in New York and another to the 701 extension in London. Now you can place secure and FREE calls between the sites by dialing 9701 from each softphone. Enjoy!
Originally published: Monday, May 2, 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.
Migrating Incredible PBX 2022 to a PUBLIC-Facing Cloud PBX
Today we want to show you how to reconfigure Incredible PBX 2022 for Rocky 8 into a PUBLIC-facing Cloud PBX. What that means is authorized users can connect a SIP phone to the PBX regardless of where the user might be located without worries about an ever-changing dynamic IP address and the requirement to whitelist the new IP address. A PUBLIC-facing PBX also provides free SIP URI connectivity to users of your PBX by anyone from anywhere in the world. In other words, it’s similar to the way you could connect to any Ma Bell telephone in the world simply by knowing the number to dial. The difference, of course, is SIP URI connectivity is free while there were often staggering long distance charges for remote connectivity in the Ma Bell days. Fifty years ago it was not uncommon for a college boy to spend $200 a month calling his college sweetheart less than 200 miles away in the same state. Ask me how I know.
Why is this such a big deal? The short answer is security and your phone bill. You don’t want bad guys on the other side of the globe attempting to register a SIP phone to your PBX so that they can use your trunks to make free phone calls on your nickel. You also don’t want anybody and everybody calling your users by simply guessing the IP address of your PBX. As with Incredible PBX 2021 PUBLIC for Debian, today’s design combines several security mechanisms to make a PUBLIC-facing PBX safe and secure. First, we will block all SIP connectivity to your PBX by IP address. Second, we will identify 30,000+ known SIP bad guys and block their access to your PBX entirely. Third, we will only permit SSH access to your PBX using public key authentication instead of traditional username/password authentication. Fourth, we will only permit web access to the Incredible PBX portal from whitelisted IP addresses and OpenVPN private addresses. We haven’t mentioned the elephant in the room, Distributed Denial of Service (DDoS) attacks, but today’s methodology reduces the risk considerably since your PBX cannot be ping’d, and all IP address access is blocked at the Linux kernel level.
Prerequisites. To put all these safeguards in place, you’ll need a cloud-based Incredible PBX 2022 KVM platform running Rocky 8. Install the latest Incredible PBX 2022 platform using our tutorial. Next, you’ll need these items:
- Public IP Address of your server
- Obscure FQDN linked to this public IP address
- Random SSH port with registered public keys for SSH access
- List of SIP extensions to enable for SIP URI access
- IP Addresses to WhiteList for Access to the Web GUI
1. Deciphering Public IP Address of Your PBX
After logging into your PBX as root, you can execute pbxstatus to decipher the public IP address of the PBX. Or issue the command: wget -q -O - ipinfo.io/ip
2. Obtaining an FQDN for Your PBX
Security through obscurity provides a critical layer of protection for your server so choose an FQDN carefully. sip.yourname.com provides little protection while f246g.yourname.com pretty much assures that nobody is going to guess your domain name. This is particularly important with SIP registrations because registered extensions on your PBX can obviously make phone calls that cost you money. If you don’t have your own domain, you can always obtain a free hostname from a service such as NoIP.com.
3. Securing SSH Access to Your PBX
Whatever you do, don’t leave SSH access via port 22 exposed on your PBX. In the time it took to create a new PBX on CloudAtCost, there were over 400 attempted logins to the default SSH port of the new server. The simplest (but least secure) method to avoid these script kiddie attacks is to change the port number for SSH access to your server. We suggest using the year you were born as the port number because it’s easy to remember. Edit /etc/ssh/sshd_config and uncomment the Port line replacing 22 with the port number you chose. Then restart SSH: systemctl restart sshd.
The preferable solution to secure SSH is to create and use SSH keys for access and set PasswordAuthentication no on the last line of /etc/ssh/sshd_config. Digital Ocean has an excellent tutorial to walk you through the setup process.
4. Choosing Extensions for SIP URI Public Access
With today’s PUBLIC design, exposing an extension for PUBLIC access means anyone in the world that knows the FQDN of your server and the extension number can do two things using any SIP client: (1) they can call you and (2) they can attempt to register to that extension and make calls on your trunks AND your nickel. So only expose extensions for public access if there is a need to connect or call from remote locations. For extensions you decide to expose, make certain that the passwords for these extensions are extremely secure, lengthy, and use numbers with both UPPER and lower case letters. Never use default extension passwords!
5. Whitelisting IP Addresses for Public Web Access
Without enumerating IP addresses for public web access, you won’t be able to connect to the web GUI of your PBX from any IP address. Down the road, if you wish to authroize additional IP addresses, you can use /root/add-ip to add them via SSH.
Deploying Incredible PBX PUBLIC Firewall
To get started, log into your server as root and issue the following commands:
cd /tmp wget http://incrediblepbx.com/newpublic.tar.gz tar zxvf newpublic.tar.gz rm -f newpublic.tar.gz
Next, edit /tmp/iptables.base and change the highlighted entries:
Change port 22 in the dport entry to the SSH port number you chose in Step 3, above.
Change 8.8.8.8, 8.8.4.4., and 1.1.1.1 to actual public IP addresses of desktop machines you wish to use to access the web GUI of your PBX. If you don’t need three entries, comment out the other entries with # at the beginning of each line.
Replace your-servers-IP-address with the actual IP address of your PBX from Step 1, above.
Save the file.
On the Rocky 8 platform, issue the following commands:
cd /etc/sysconfig cp /tmp/iptables.base . mv iptables iptables.orig cp iptables.base iptables
Using Incredible PBX PUBLIC with Asterisk
The first line of defense with this PUBLIC implementation is your FQDN. Second is the IPtables firewall setup above. And third is the Asterisk® extensions configuration in extensions_override_freepbx.conf. Here’s how to configure it. Edit /tmp/extensions_override_freepbx.base and change the highlighted entries:
If there are phone numbers assigned to your PBX that you want processed according to your Inbound Routing rules, duplicate the first highlighted line above and, for each trunk, replace 8881234567 with your actual DID numbers.
In exten => _.,1 line, replace your-servers-IP-address with the actual IP address of your PBX from Step 1, above.
In exten => _.,10 line, replace your-servers-FQDN with the actual FQDN assigned to your PBX from Step 2, above.
Scroll down in the file to the following section:
Comment out undesired default extensions. Place a semicolon at the beginning of the lines.
For any extensions you wish to add, insert a new line in the following format replacing both 7000 entries with the desired extension number:
exten => 7000,13,Dial(local/7000@from-internal)
Save the file and then execute the following commands to complete the PUBLIC setup:
cd /etc/asterisk cp /tmp/extensions_override_freepbx.base . mv extensions_override_freepbx.conf extensions_override_freepbx.orig cp extensions_override_freepbx.base extensions_override_freepbx.conf fwconsole restart asterisk -rx "dialplan reload" iptables-restart sed -i 's|-A INPUT|-I INPUT|' /root/add-ip sed -i 's|-A INPUT|-I INPUT|' /root/add-fqdn
Adding IPSET Incredible PBX 2022 Protection
We’re not the biggest fans of blacklists because the bad guys spend a lot of time trying to corrupt them by inserting valid IP addresses of sites such as DNS servers in the lists to wreak havoc. Having said that, there are two blacklists that are carefully monitored on a daily basis, and both provide additional protection for your PBX by weeding out access by 30,000+ potential bad guys. The oldest of these is VoIP Blacklist. And the new kid on the block is APIBAN from LOD.com and Fred Posner. We’ve simplified the setup process for use with Incredible PBX 2022. To get started, obtain an APIBAN API key here. Then issue the following commands to put all the pieces in place on your server:
cd /usr/local/sbin wget http://incrediblepbx.com/incrediblepbx-ipsets8.tar.gz tar zxvf incrediblepbx-ipsets8.tar.gz rm -f incrediblepbx-ipsets8.tar.gz
Next, edit /usr/local/sbin/apiban-init and insert your APIkey.
Finally, issue the following command to reload the firewall: iptables-restart
Verifying Firewall Setup of Incredible PBX
Let’s make certain that everything got installed correctly. Begin by issuing this command: iptables -nL
Scroll toward the top of the list, and you should see two entries for the voipbl and apiban ipsets indicating that entries in those lists will be dropped by the firewall.
Next, verify that the voipbl and apiban ipsets are populated. The first two commands below will list all of the blocked IP addresses. And the next two commands will provide a count of the dropped IP addresses.
ipset list voipbl ipset list apiban ipset list voipbl | wc -l ipset list apiban | wc -l
Finally, you can refresh the ipsets with the following two commands:
voipbl-init apiban-init
Rebooting or restarting the firewall with iptables-restart also refreshes the ipset listings.
Calling an Incredible PBX PUBLIC Extension
Any extensions that you have whitelisted in the blue section above can be called from anywhere using any SIP client. Simply enter the SIP URI for the extension in the following format: SIP/extension@your-servers-FQDN
CAUTION: If a caller attempts to call any extension on your PUBLIC server from an extension on another Asterisk server to which the caller is registered, the call will fail if there is a matching extension number on the PUBLIC server and the two servers are not registered to each other. So remember to use unique extension numbers on your PUBLIC server if you expect callers from other Asterisk servers.
Registering Incredible PBX PUBLIC Extension
If you wish to login to a whitelisted extension using a SIP client, enter the extension and password of the extension. For the server address, enter the FQDN of your server. If it’s a PJsip extension, add :5061 to the end of the FQDN.
Originally published: Monday, April 25, 2022
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
5 Minute Wonder: Incredible PBX 2022 in Cloud for $25/Yr.
We’ve been building turnkey Incredible PBX® servers for virtual machine platforms for many years. Because the servers are built from images, typical install times have been 5 minutes or less with Proxmox, VirtualBox, and VMware ESXi. But the missing piece has been a similar image install on a true cloud-based platform. This limitation was primarily due to the fact that we don’t own or control the available cloud platforms which typically limit image installs to operating systems such as CentOS, Debian, Ubuntu, and Windows. However, CrownCloud was good enough to add our Incredible PBX 2021 Debian image and the Incredible PBX 2020 CentOS 7 image to their portfolio. And, today, we have more good news. CrownCloud has now added the Incredible PBX 2022 image for Rocky 8 as well.
These 5-minute turnkey installs of Incredible PBX 2020 for CentOS 7, Incredible PBX 2021 for Debian 10, and now Incredible PBX 2022 for Rocky 8 are being offered at the jaw-dropping price of $25 a year. The monthly cost is cheaper than a cup of coffee at Starbucks, and you’ll have a fully-functioning, production-ready KVM platform including a free snapshot with 1GB RAM, 20GB SSD storage, and 1TB of monthly bandwidth in your choice of server locations including Los Angeles and Atlanta in the United States as well as Germany and the Netherlands in Europe. And, unlike all of the other Asterisk® aggregations, Incredible PBX still provides a source code-based platform that can be tailored to meet any special requirements your organization may need.
We don’t make a nickel on these offerings so consider this our special thanks to all of our loyal fans.
Here are the links to sign up for the service and take advantage of these Incredible PBX deals:
- Los Angeles: https://crowncloud.net/clients/cart.php?a=add&pid=382
- Atlanta: https://crowncloud.net/clients/cart.php?a=add&pid=487
- Germany: https://crowncloud.net/clients/cart.php?a=add&pid=399
- Netherlands: https://crowncloud.net/clients/cart.php?a=add&pid=400
When you sign up for the service, choose any traditional OS for the base install. Once it’s on line, go into CrownPanel and choose Reinstall. Then select Incredible PBX 2022 for Rocky from the Application Images pulldown as your new install. In less than 5 minutes, your server will be ready for you to login. Be sure to use SSH and NOT the VNC utility included in CrownPanel. This will assure that your desktop machine’s IP address gets whitelisted in the Incredible PBX firewall. Otherwise, you won’t be able to SSH into your server from your desktop. Once you login, the Incredible PBX configurator will prompt you to set passwords for root login, admin login to FreePBX, and admin login credentials for Apache to access AsteriDex and Reminders. Add a trunk provider (Skyetel is preconfigured and enabled by default) and a softphone or Incredible PBX SIP phone, and your PBX is fully operational. Check out the CrownCloud Wiki.
Planning Ahead for That Rainy Day
One of our favorite features of Crown Cloud is the free snapshot (a.k.a. backup) at no additional charge. We recommend you take snapshots regularly as you make major changes in your server’s configuration. In this way, if something comes unglued, you can easily restore the snapshot and never miss a beat. You’ll find the Remote Snapshot option in your CrownPanel menu.
Configuring Skyetel for Incredible PBX
If you’ve decided to go with Skyetel, here’s the drill. Sign up for Skyetel service and take advantage of the Nerd Vittles specials. First, complete the Prequalification Form here. You then will be provided a link to the Skyetel site to complete your registration. Once you have registered on the Skyetel site and your account has been activated, open a support ticket and request the $10 credit for your account by referencing the Nerd Vittles special offer. Once you are satisfied with the service, fund your account as desired, and Skyetel will match your deposit of up to $250 simply by opening another ticket. That gets you up to $500 of half-price calling. Credit is limited to one per person/company/address/location. Effective 10/1/2023, $25/month minimum spend required.
Skyetel does not require SIP registrations to make connections to your PBX. Instead, Skyetel can use Endpoint Groups to identify which servers can communicate with the Skyetel service. An Endpoint Group consists of a Name, an IP address, a UDP or TCP port for the connection, and a numerical Priority for the group. For incoming calls destined to your PBX, DIDs are associated with an Endpoint Group to route the calls to your PBX. For outgoing calls from your PBX, a matching Endpoint Group is required to authorize outbound calls through the Skyetel network. Thus, the first step in configuring the Skyetel side for use with your PBX is to set up an Endpoint Group. Here’s a typical setup for Incredible PBX:
- Name: MyPBX
- Priority: 1
- IP Address: PBX-Public-IP-Address
- Port: 5060
- Protocol: UDP
- Description: my.incrediblepbx.com
To receive incoming PSTN calls, you’ll need at least one DID. On the Skyetel site, you acquire DIDs under the Phone Numbers tab. You have the option of Porting in Existing Numbers (free for the first 60 days after you sign up for service) or purchasing new ones under the Buy Phone Numbers menu option.
Once you have acquired one or more DIDs, navigate to the Local Numbers or Toll Free Numbers tab and specify the desired SIP Format and Endpoint Group for each DID. Add SMS/MMS and E911 support, if desired. Call Forwarding and Failover are also supported. That completes the VoIP setup on the Skyetel side. System Status is always available here. Everything is already in place on the Incredible PBX 2022 side of the house so you can start making and receiving calls immediately.
Configuring ClearlyIP SIP Trunking
For the tightest integration with FreePBX, no SIP provider holds a candle to Incredible PBX SIP Trunking with ClearlyIP. The reason is fairly obvious. The ClearlyIP folks were the original developers of FreePBX. In addition to all of the traditional SIP trunking services, you also get CNAM support and state-of-the-art E911 service which can be deployed in full compliance with Kari’s Law and the Ray Baum Act. If you’re a system integrator and don’t know about your financial liability for failure to comply with the new rules, it’s time to do some reading.
Configuring VoIP.ms for Incredible PBX
To sign up for VoIP.ms service, may we suggest you use our signup link so that Nerd Vittles gets a referral credit for your signup. Once your account is set up, you’ll need to set up a SIP SubAccount and, for Authentication Type, choose Static IP Authentication and enter your Incredible PBX server’s public IP address. For Transport, choose UDP. For Device Type, choose Asterisk, IP PBX, Gateway or VoIP Switch. Order a DID in their web panel, and then point the DID to the SubAccount you just created. Be sure to specify atlanta1.voip.ms as the POP from which to receive incoming calls. In the Incredible PBX GUI, be sure to enable the VoIP.ms trunk.
Configuring Anveo Direct for Incredible PBX
To sign up for Anveo Direct service, sign up on their web site and then login. After adding funds to your account, purchase a DID under Inbound Service -> Order DID. Next, choose Configure Destination SIP Trunk. Give the Trunk a name. For the Primary SIP URI, enter $[E164]$@server-IP-address. For Call Options, select your new DID from the list. You also must whitelist your public IP address under Outbound Service -> Configure. Create a new Call Termination Trunk and name it to match your server. For Dialing Prefix, choose six alphanumeric characters beginning with a zero. In Authorized IP Addresses, enter the public IP address of your server. Set an appropriate rate cap. We like $0.01 per minute to be safe. Set a concurrent calls limit. We like 2. For the Call Routing Method, choose Least Cost unless you’re feeling extravagant. For Routes/Carriers, choose Standard Routes. Write down your Dialing Prefix and then click the Save button.
Before you can make outbound calls through Anveo Direct from your PBX, you first must configure the Dialing Prefix that you wrote down in the previous step. Log into the GUI as admin using a web browser and edit the Anveo-Out trunk in Connectivity -> Trunks. Enable the Trunk. Then click on the custom-Settings tab and replace anveo-pin with your actual Dialing Prefix. Click Submit and Apply Config to complete the setup. In the Incredible PBX GUI, be sure to enable all of the remaining Anveo trunks.
By default, incoming Anveo Direct calls will be processed by the Default inbound route on your PBX. If you wish to redirect incoming Anveo Direct calls using DID-specific inbound routes, then you’ve got a bit more work to do. In addition to creating the inbound route using the 11-digit Anveo Direct DID, enter the following commands after logging into your server as root using SSH/Putty:
cd /etc/asterisk
echo "[from-anveo]" >> extensions_custom.conf
echo "exten => _.,1,Ringing" >> extensions_custom.conf
echo "exten => _.,n,Goto(from-trunk,\\${SIP_HEADER(X-anveo-e164)},1)" >> extensions_custom.conf
asterisk -rx "dialplan reload"
Configuring a SIP Phone for Incredible PBX 2022
We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which is a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. We obviously recommend the Incredible PBX IP phones from ClearlyIP which are the most versatile.
If you’ve been keeping up with recent Nerd Vittles developments, then you already know that we have just introduced a new Cellular Phone which connects directly to your PBX and serves as a perfect remote extension and traveling companion. You can read all about it here.
Software devices such as the YateClient softphone are another option for desktop machines. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the VoIP-Info.org Forum when you’re ready to get serious about VoIP telephony.
We recommend YateClient for Windows which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the 701 extension on Incredible PBX. You can find them by running /root/show-passwords. You’ll need the IP address of your server plus your extension 701 password. In the YateClient, fill in the blanks using the IP address of your Server plus :5061 for the PJsip 701 extension, 701 for your Username, and whatever Password was assigned to the extension when you installed Incredible PBX. Click OK to save your entries.
Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls to the numerous apps that are preconfigured on Incredible PBX. Dial a few of these to get started:
DEMO - Apps Demo 123 - Reminders 947 - Weather by ZIP Code 951 - Yahoo News TODAY - Today in History LENNY - The Telemarketer's Worst Nightmare
If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store. For Android users, check out the terrific new VitalPBX Communicator. Works flawlessly with Incredible PBX.
For smartphone solutions, visit the Incredible PBX Wiki for our softphone recommendations.
Configuring SendMail with Incredible PBX
In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. We strongly recommend configuring SendMail using either your ISP or Gmail as an SMTP Relay Host. NOTE: If you are using a Gmail account with 2-step verification enabled, you MUST use a Gmail App Key instead of your Gmail account password. You also must enable Less Secure Apps access to the Gmail account.
Configuring a Gmail account with Incredible PBX 2022 is as simple as entering your Gmail credentials. Just run this script: /root/enable-gmail-smarthost-for-sendmail.
Here are the steps using a Gmail account with Incredible PBX 2020:
cd /etc/mail yum -y install sendmail-cf hostname -f > genericsdomain touch genericstable cd /usr/bin rm -f makemap ln -s ../sbin/makemap.sendmail makemap cd /etc/mail makemap -r hash genericstable.db < genericstable mv sendmail.mc sendmail.mc.original wget http://incrediblepbx.com/sendmail.mc.gmail cp sendmail.mc.gmail sendmail.mc mkdir -p auth chmod 700 auth cd auth echo AuthInfo:smtp.gmail.com \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" > client-info echo AuthInfo:smtp.gmail.com:587 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info echo AuthInfo:smtp.gmail.com:465 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info # Stop here and edit client-info (nano -w client-info) in all three lines. # Replace user_id with your gMail account name without @gmail.com # Replace password with your real gMail password OR # use your Gmail App Key if 2-step verification is enabled # Be sure to replace the double-quotes shown above if they don't appear in the file!!! # Save your changes (Ctrl-X, Y, then Enter) chmod 600 client-info makemap -r hash client-info.db < client-info cd .. make systemctl restart sendmail
Even though these servers are hosted in the cloud, we still recommend using a SmartHost to minimize email delivery problems.
Test outbound mail using this command with your actual email address:
echo "test" | mail -s testmessage yourname@youremaildomain.com
On some implementations, you may notice in the FreePBX GUI that the mail queue has failed. Here's the fix:
chmod 777 /var/spool/mqueue service sendmail restart
Once you are sure your emails are being delivered reliably, here's a sample GUI voicemail configuration for an extension:
Be advised that Google has hinted that the Gmail Smarthost landscape may be changing. See our recent article for a simple SmartHost alternative.
Incredible PBX Administration
We've eased the pain of administering your new PBX with a collection of scripts which you will find in the /root folder after logging in with SSH or Putty. Here's a quick summary of what each of the scripts does.
add-fqdn is used to whitelist a fully-qualified domain name in the firewall. Because Incredible PBX blocks all traffic from IP addresses that are not whitelisted, this is what you use to authorize an external user for your PBX. The advantage of an FQDN is that you can use a dynamic DNS service to automatically update the IP address associated with an FQDN so that you never lose connectivity.
add-ip is used to whitelist a public IP address in the firewall. See the add-fqdn explanation as to why this matters.
del-acct is used to remove an IP address or FQDN from the firewall's whitelist.
admin-pw-change is used to set the admin password for access to the FreePBX/Incredible PBX web GUI using a browser pointed to the local IP address of your server.
apache-pw-change is used to set the admin password for access to Apache/Incredible PBX apps including AsteriDex and Reminders. This provides a password layer of protection for access to these applications.
reset-conference-pins is a script that automatically and randomly resets the user and admin pins for access to the preconfigured conferencing application. Dial C-O-N-F from any registered SIP phone to connect to the conference.
reset-extension-passwords is a script that automatically and randomly resets ALL of the SIP passwords for extensions 702-705. Be careful using this one, or you may disable existing registered phones and cause Fail2Ban to blacklist the IP addresses of those users. HINT: You can place a call to the Ring Group associated with all five extensions by dialing 777.
reset-reminders-pin is a script that automatically and randomly resets the pin required to access the Telephone Reminders application by dialing 123. It's important to protect this application because a nefarious user could set up a reminder to call a number anywhere in the world assuming your SIP provider's account was configured to allow such calls.
show-feature-codes is a cheat sheet for all of the feature codes which can be dialed from any registered SIP phone. It documents how powerful a platform Incredible PBX actually is. A similar listing is available in the GUI at Admin -> Feature Codes.
show-passwords is a script that displays most of the passwords associated with Incredible PBX. This includes SIP extension passwords, voicemail pins, conference pins, telephone reminders pin, and your Anveo Direct outbound calling pin (if configured). Note that voicemail pins are configured by the user of a SIP extension the first time the user accesses the voicemail system by dialing *97.
ssh-regen.sh allows you to reset the SSH keys for your server for added security.
update-IncrediblePBX is the Automatic Update Utility which checks for server updates from incrediblepbx.com every time you log into your server as root using SSH or Putty. Do NOT disable it as it is used to load important fixes and security updates when necessary. We recommend logging into your server at least once a week.
pbxstatus (shown above) displays status of all major components of Incredible PBX.
Forwarding Calls to Your Cellphone. Keep in mind that inbound calls to your DIDs automatically ring all five SIP extensions, 701-705. The easiest way to also ring your cellphone is to set one of these five extensions to forward incoming calls to your cellphone. After logging into your PBX as root, issue the following command to forward calls from extension 705 to your cellphone: asterisk -rx "database put CF 705 6781234567"
To remove call forwarding: asterisk -rx "database del CF 705"
Keeping FreePBX 15 Modules Current
We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:
rm -f /tmp/* fwconsole ma upgradeall fwconsole reload /root/sig-fix /root/sig-fix
Where To Go From Here
Complete documentation on the ClearlyIP Devices Module is available here.
Complete documentation on the FreePBX GPL Modules is available here.
Complete documentation on the Incredible PBX additions is available here.
An introduction to configuring extensions, trunks, and routes is available here.
Free voicemail transcription with email delivery. Tutorial available here.
Setting Up a VPN for Your PBX: OpenVPN or NeoRouter
Originally published: Monday, March 28, 2022
Need help with Asterisk? Visit the VoIP-info Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Deploying a Non-Google SMTP RelayHost with Asterisk
This will not be the sexiest column you read this year, but it may be the most important. You need a reliable way for your VoIP-based PBX to deliver emails to you and your users when incoming voicemails arrive and when your server has problems. At least for Incredible PBX® platforms, we thought we had this solved with our Gmail Smarthost solution. But, alas, Google continues to move the goal posts and has announced that it will discontinue support for so-called "Less Secure Apps" on May 30, 2022. After watching Google blow up one of their apps after another over the years, you’d think we’d learn. But the ease of use and (free) pricing of Google offerings continues to entice until… another one bites the dust.
Why Do You Need a RelayHost? Many Internet Service Providers (ISPs) block downstream mail servers from sending email to reduce spam. If you have a PBX sitting behind a Comcast cable modem and you don’t have a business account, that would be you.
What Is a RelayHost? A RelayHost is an intermediate mail provider that provides last mile delivery of your server’s outbound email without your having to worry about the intricacies of setting up and properly configuring an SMTP gateway. Instead, your server pushes your outbound email messages to the smarthost using your credentials and leaves the rest of the delivery task to the RelayHost.
Choosing a RelayHost. Lucky for all of us, there are many RelayHost providers from which to choose. Most offer a free tier with 100 or more daily emails. For most PBXs, that is more than ample without spending a dime. If your server pushes out more than 100 emails a day, then there are commercial tiers of service available from these same providers. Here are some of the favorites recommended by our users: Twilio’s SendGrid, Mailjet, SendInBlue, and our own free MXroute offering. We would prefer you use our service as a last resort if you can’t get any of the other free offerings to work. We’ll show you how to set up Twilio’s SendGrid as a RelayHost to get you started.
Configuring a RelayHost with Incredible PBX. There’s good news and bad news. While all of these solutions offer a free tier, the setup process with some of these services can be a bear. We’ve attempted to take the pain out of this by walking you through the setup steps. For openers, configuring SendMail as an SMTP Smarthost is not for mere mortals so we’ll first migrate your PBX to Postfix to simplify the setup procedure. Here is a quick list of the tasks:
- Migrate Your PBX from SendMail to Postfix
- Obtain an Account with SmartHost Provider
- Configure Postfix Email Relay Using SmartHost Provider
- Test Outbound Mail
Migrating Incredible PBX 2022 to Postfix. Our new Incredible PBX 2022 build for Rocky 8 comes with both SendMail and Postfix preinstalled. So it’s simple to switch gears. Here are the commands:
systemctl stop sendmail systemctl disable sendmail systemctl start postfix systemctl enable postfix sed -i 's|-c sendmail|-c postfix|' /usr/local/sbin/pbxstatus sed -i 's|SendMail| Postfix|' /usr/local/sbin/pbxstatus systemctl status postfix
Migrating Incredible PBX 2021 to Postfix. By default, Incredible PBX 2021 servers do not include Postfix. So here are the steps to install Postfix and remove SendMail. These steps also apply to all releases of Incredible PBX on the Raspbian platform.
apt-get update apt-get install postfix sasl2-bin -y # choose No Configuration option when prompted in: # dpkg-reconfigure postfix systemctl stop sendmail systemctl disable sendmail systemctl start postfix systemctl enable postfix sed -i 's|-c sendmail|-c postfix|' /usr/local/sbin/pbxstatus echo "incrediblepbx.com" > /etc/mailname sed -i 's|SendMail| Postfix|' /usr/local/sbin/pbxstatus systemctl status postfix
Creating a Free SendGrid Account. Begin by navigating to the SendGrid Start for Free link. Enter your Email Address and a 16-character password of your choice. Accept the Terms of Service and click Create Account. Fill in the Personal Information and click Get Started. Create a Single Sender Identity. Confirm receipt of the verification email. You can skip enabling Two-Factor Authentication if desired. Next, open the Email API tab and click Integration Guide. Choose SMTP Relay and create an APIKEY. Copy the generated APIKEY to a safe place. It cannot be deciphered again!
Now edit /etc/postfix/main.cf and add the following entries to the bottom of the file:
smtp_sasl_auth_enable = yes smtp_sasl_password_maps = hash:/etc/postfix/sasl_passwd smtp_sasl_security_options = noanonymous smtp_sasl_tls_security_options = noanonymous smtp_tls_security_level = encrypt header_size_limit = 4096000 relayhost = [smtp.sendgrid.net]:587
Then create /etc/postfix/sasl_passwd and enter the following, replacing YOUR-APIKEY with your actual key from above:
[smtp.sendgrid.net]:587 apikey:YOUR-APIKEY
Complete the setup by issuing the following commands:
chmod 600 /etc/postfix/sasl_passwd postmap /etc/postfix/sasl_passwd systemctl restart postfix systemctl status postfix
IMPORTANT NOTE: With SendGrid, the email address you entered for your Single Sender Identity must match the FROM: address on every outbound email message sent from your server. We’ll need to adjust the FROM: address in FreePBX before any voicemail emails can be successfully sent. Also, if you send emails from the command line, the syntax must be as shown here with your sender@yourdomain.com matching what was entered as your Sender Identity email address.
echo "test" | mail -r sender@yourdomain.com -s testmessage recipient@somedomain.com
At least with Gmail recipients, they may also see the following with messages from your PBX until they click Looks Safe:
Finally, be sure to adjust the FROM address for outbound voicemail messages in the FreePBX GUI. Login as admin and enter your Sender Identity Email Address in the Server Email field in Settings -> Voicemail Admin -> Settings -> Email Config.
You can check for errors by reviewing the Postfix mail log: tail /var/log/mail.log.
UPDATE: Setup scripts for Debian 10 and Rocky 8 can be downloaded here.
Originally published: Thursday, March 17, 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.
One-Minute Wonder: It’s Incredible PBX 2022 for VirtualBox
If you’re new to the VoIP world and want to kick the tires to see what you’re missing, then today’s one minute setup is for you. You can use almost any desktop computer you already own to bring up the VirtualBox® edition of Incredible PBX® 2022.
If you’ve followed Nerd Vittles over the years, you already know that VirtualBox from Oracle® is one of our favorite platforms. Once VirtualBox is installed on your desktop computer, adding Incredible PBX is a snap. Download the latest Incredible PBX 2022 image from SourceForge, double-click on the downloaded image, check the initialize MAC address box, and boom. In less than a minute, your PBX is ready to use with the very latest Rocky 8 platform and Asterisk® 18 build plus all of the FreePBX® 15 GPL modules. There are no hidden fees or crippleware to hinder your use of Incredible PBX for as long as you like. If you set up an account with our Platinum provider, Skyetel, you can start making calls in minutes. Of course, the Incredible PBX feature set is included as well which brings you nearly three dozen applications for Asterisk® that will revolutionize your communications platform. Speech-to-text, voice recognition, and a Siri-like telephony interface are as close as your SIP phone.
Installing Oracle VM VirtualBox
Oracle’s virtual machine platform inherited from Sun is amazing. It’s not only free, but it’s pure GPL2 code. VirtualBox gives you a virtual machine platform that runs on top of any desktop operating system. In terms of limitations, we haven’t found any. We even tested this on an Atom-based Windows 7 machine with 2GB of RAM, and it worked without a hiccup. So step #1 today is to download one or more of the VirtualBox installers from VirtualBox.org or Oracle.com. Our recommendation is to put all of the 100MB installers on a 4GB thumb drive.1 Then you’ll have everything in one place whenever and wherever you happen to need it. Once you’ve downloaded the software, simply install it onto your favorite desktop machine. Accept all of the default settings, and you’ll be good to go. For more details, here’s a link to the Oracle VM VirtualBox User Manual.
NOTE: The Incredible PBX 2022 VM requires a VirtualBox 6.x platform. Adjust screen size in View -> Virtual Screen.
Installing the Incredible PBX 2022 Image
To begin, download the Incredible PBX 2022 image (3.5 GB) onto your desktop.
Next, double-click on the Incredible PBX .ova image on your desktop. Be sure to check the box to initialize the MAC address of the image if you’re using an older version of VirtualBox. Then click Import. Once the import is finished, you’ll see a new Incredible PBX 2022 virtual machine in the VM List of the VirtualBox Manager Window. Let’s make a couple of one-time adjustments to the Incredible PBX configuration to account for possible differences in sound and network cards on different host machines.
(1) Click once on the Incredible PBX virtual machine in the VM List. Then (2) click the Settings button. In System tab, verify Hardware Clock in UTC Time is checked. In the Audio tab, check the Enable Audio option and choose your sound card. In the Network tab for Adapter 1, check the Enable Network Adapter option. From the Attached to pull-down menu, choose Bridged Adapter. Then select your network card from the Name list. Then click OK. That’s all the configuration that is necessary for Incredible PBX 2022.
Running Incredible PBX 2022 in VirtualBox
Once you’ve imported and configured the Incredible PBX 2022 Virtual Machine, you’re ready to go. Highlight the Incredible PBX 2022 virtual machine in the VM List on the VirtualBox Manager Window and click the Start button. The standard Linux boot procedure will begin and, within a few seconds, you’ll get the familiar Linux login prompt. During the bootstrap procedure, you’ll see a couple of dialogue boxes pop up that explain the keystrokes to move back and forth between your host operating system desktop and your virtual machine. Remember, you still have full access to your desktop computer. Incredible PBX is merely running as a task in a VM window. Always gracefully halt Incredible PBX just as you would on any computer.
Here’s what you need to know. To work in the Incredible PBX virtual machine, just left-click your mouse while it is positioned inside the VM window. To return to your host operating system desktop, press the right Option key on Windows machines or the left Command key on any Mac. On Linux desktops, press the right Ctrl key. For other operating systems, read the dialogue boxes for instructions on moving around. To access the Linux CLI, login as root with the default password: password. Change your root password when you are prompted to do so. Then update your admin password for web access: ./admin-pw-change. Also update your admin password for web applications: ./apache-pw-change. You’ll need these admin passwords to access the web GUI to manage your PBX as well as to use the AsteriDex and Reminders web apps. The above password updates are automatically requested when you first activate the virtual machine. You can update all of your other passwords using the scripts provided in /root.
Setting the Date and Time with VirtualBox
On some platforms, VirtualBox has a nasty habit of mangling the date and time of your virtual machine. Verify that you have enabled the Hardware Clock in UTC Time option for your virtual machine as documented above. If pbxstatus still shows an incorrect time, manually set the date and time and then update the hardware clock. Here’s how assuming 08130709 is the month (August), day (13), and correct time (7:09 a.m.) of your server:
date 08130709 clock -w
Configuring Skyetel for Incredible PBX 2022
If you’d like to try out the Skyetel service at no charge, here’s the drill. Sign up for Skyetel service to 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, and address. 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 2022:
- Name: MyPBX
- Priority: 1
- IP Address: PBX-Public-IP-Address
- Port: 5061
- 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.
If VirtualBox is sitting behind a router or firewall on a private LAN, you’ll need to forward ports UDP 5060, 5061, and 10000-20000 in your router to the private LAN address of your Incredible PBX server. Also edit your extensions in the GUI and set NAT=YES in the Advanced tab of every extension. In Settings -> Asterisk SIP Settings, click the Detect Network Settings button and then Submit your changes and reload the Asterisk dialplan when prompted.
Finally, login to the FreePBX web GUI as admin using the password you assigned when you set up the virtual machine. Navigate to Connectivity -> Trunks and edit the Skyetel-pjSIP trunk. Change the Disable Trunk setting from Yes to No. Then click Submit and reload your dialplan when prompted. That’s it.
Configuring VoIP.ms for Incredible PBX 2022
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 2020 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. For more details about VoIP.ms, see this Nerd Vittles tutorial.
Configuring SendMail with Incredible PBX 2022
In order to receive voicemails by email delivery, outbound mail functionality from your server obviously is required. If you’ve deployed your server in your home, your Internet Service Provider probably blocks downstream mail servers such as Incredible PBX from sending mail. This is done to reduce SPAM. In this case, you will need to configure SendMail using either your ISP or Gmail as an SMTP Relay Host. We have built aninstall script to set up a SmartHost using Gmail. Simply run it and insert your Gmail username and password or App Password.
cd /root wget http://incrediblepbx.com/enable-gmail-smarthost-rocky8.tar.gz tar zxvf enable-gmail-smarthost-rocky8.tar.gz rm -f enable-gmail-smarthost-rocky8.tar.gz ./enable-gmail-smarthost-for-sendmail
Configuring a Softphone for Incredible PBX 2022
We’re in the home stretch now. You can connect virtually any kind of telephone to your new PBX. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as the Incredible PBX SIP phone. 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 in the Incredible PBX Wiki 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 navigating to Applicaations -> Extensions -> 701 in the FreePBX GUI.
Configuring Incredible PBX 2022 for VirtualBox
In order to take advantage of all the Incredible PBX applications, you’ll need to obtain IBM text-to-speech (TTS) and speech-to-text (STT) credentials as well as a (free) Application ID for Wolfram Alpha.
This Nerd Vittles tutorial will walk you through getting your IBM account set up and obtaining both your TTS and STT credentials. Be sure to write down BOTH sets of credentials which you’ll need in a minute. For home and SOHO use, IBM access and services are mostly FREE even though you must provide a credit card when signing up. The IBM signup process explains their pricing plans.
To use Wolfram Alpha, sign up for a free Wolfram Alpha API account. Just provide your email address and set up a password. It takes less than a minute. Log into your account and click on Get An App ID. Make up a name for your application and write down (and keep secret) your APP-ID code. That’s all there is to getting set up with Wolfram Alpha. If you want to explore costs for commercial use, there are links to let you get more information.
In addition to your Wolfram Alpha APPID, there are two sets of IBM credentials to plug into the Asterisk AGI scripts. Keep in mind that there are different usernames and passwords for the IBM Watson TTS and STT services. The TTS credentials will look like the following: $IBM_username and $IBM_password. The STT credentials look like this: $API_USERNAME and $API_PASSWORD. Don’t mix them up. 🙂
All of the scripts requiring credentials are located in /var/lib/asterisk/agi-bin so switch to that directory after logging into your server as root. Edit each of the following files and insert your TTS credentials in the variables already provided: nv-today2.php, ibmtts.php, and ibmtts2.php. Edit each of the following files and insert your STT credentials in the variables already provided: getquery.sh, getnumber.sh, and getnumber2.sh. Finally, edit 4747 and insert your Wolfram Alpha APPID.
Using AsteriDex with Incredible PBX
AsteriDex is a web-based dialer and address book application for Asterisk and Incredible PBX. It lets you store and manage phone numbers of all your friends and business associates in an easy-to-use SQLite3 database. You simply call up the application with your favorite web browser: http://pbx-ip-address/asteridex4/. When you click on a contact that you wish to call, AsteriDex first calls you at extension 701, and then AsteriDex connects you to your contact through another outbound call made using your default outbound trunk that supports numbers in the 1NXXNXXXXXX format.
Keeping FreePBX 15 Modules Current
We strongly recommend that you periodically update all of your FreePBX modules to eliminate bugs and to reduce security vulnerabilities. From the Linux CLI, log into your server as root and issue the following commands:
rm -f /tmp/* fwconsole ma upgradeall fwconsole reload /root/sig-fix /root/sig-fix
Taking Incredible PBX 2022 for a Test Drive
You can take Incredible PBX on a test drive by dialing D-E-M-O (3366) from any phone connected to your PBX.
With Allison’s Demo IVR, you can choose from the following options:
- 0. Chat with Operator — connects to extension 701
- 1. AsteriDex Voice Dialer – say "Delta Airlines" or "American Airlines" to connect
- 2. Conferencing – log in using 1234 as the conference PIN
- 3. Wolfram Alpha Almanac – say "What planes are flying overhead"
- 4. Lenny – The Telemarketer’s Worst Nightmare
- 5. Today’s News Headlines — courtesy of Yahoo! News
- 6. Weather by ZIP Code – enter any 5-digit ZIP code for today’s weather
- 7. Today in History — courtesy of OnThisDay.com
- 8. Chat with Nerd Uno — courtesy of SIP URI connection to 3CX iPhone Client
- 9. DISA Voice Dialer — say any 10-digit number to be connected
- *. Current Date and Time — courtesy of Incredible PBX
We Missed You During February
If you missed us last month, we missed you, too. We took a brief timeout to get some new eyeballs. Ah, the miracles of modern medicine. As the old song says, "I Can See Clearly Now." It’s been 35 years since I saw the world without the need for glasses. It’s good to be back.
Originally published: Monday, March 7, 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.
- Many of our purchase links refer users to Amazon when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from Amazon to help cover the costs of our blog. We never recommend particular products solely to generate Amazon commissions. However, when pricing is comparable or availability is favorable, we support Amazon because Amazon supports us. [↩]
Introducing Incredible PBX 2022 for Rocky 8 Linux
As we celebrate the 17th birthday of Nerd Vittles this week, it seemed only fitting to introduce a new version of Incredible PBX 2022 which will carry telephony forward until the end of this decade. For history buffs, you may recall that CentOS used to be a perfect clone of Red Hat Enterprise Linux… until it wasn’t when RedHat decided to morph CentOS into a development sandbox known as CentOS Stream. And then along came Gregory Kurtzer, the original developer of CentOS, to work his magic once again with the Rocky 8 reincarnation. Until Rocky 8 rose from the ashes, we had all but decided to end our CentOS distribution and focus on Debian, Raspbian, and Ubuntu. But the appeal of a new operating system with an end of life slated for May 2029 was hard to resist. So here we are.
Our roadmap for Incredible PBX 2022 for Rocky 8 goes like this. Today we are introducing this beta featuring Asterisk® 16 and the GPL modules from FreePBX® 15. Once we get the kinks out of this release, we will add two additional versions, one featuring Asterisk 18 and FreePBX 15 plus a second one featuring Asterisk 19 and FreePBX 16. The PHP platform to support all three versions is already in place. It supports both PHP 5.6 and PHP 7.4 running concurrently. We needed PHP 5.6 to preserve the legacy AvantFax application which provides free faxing.
CAUTION: Because of frequent changes to the Rocky 8 platform, we no longer recommend it for production use with Incredible PBX.
We hope you’ll download and install Rocky 8 as well as Incredible PBX 2022 and help us move this platform forward. 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 our Platinum sponsor, Vultr. They host both Rocky 8 and AlmaLinux 8 images in addition to other operating systems so platform setup takes only a minute.
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 wget http://incrediblepbx.com/IncrediblePBX2022.16.15-4.sh chmod +x IncrediblePBX2022.16.15-4.sh ./IncrediblePBX2022.16.15-4.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 ./IncrediblePBX2022.16.15-4.sh
1/31/2022 UPDATE: We’ve now released the second iteration of Incredible PBX 2022 for Rocky8 and AlmaLinux8 featuring Asterisk 18 and the FreePBX GPL modules. You can install this version as an alternative to the Asterisk 16 release if you would prefer:
cd /root yum -y install wget wget http://incrediblepbx.com/IncrediblePBX2022.18.15-1.sh chmod +x IncrediblePBX2022.18.15-1.sh ./IncrediblePBX2022.18.15-1.sh
Once the Phase I install finishes, your server will reboot. Log back in as root and run the new installer a second time:
cd /root ./IncrediblePBX2022.18.15-1.sh
If you chose to install AlmaLinux 8, issue the following command once you’ve logged in to the Linux CLI as root:
sed -i 's|Rocky 8|AlmaLinux 8|' /usr/local/sbin/pbxstatus
As with all Incredible PBX builds, a preconfigured Linux firewall is included which blocks all access except from whitelisted IP addresses. To activate it, simply press the ENTER key when the install completes. To load the latest updates to the platform using the Automatic Update Utility, simply log out of your server and log back in as root. You’re now ready to explore the Incredible PBX feature set using the tutorials available on the Incredible PBX Wiki.
Audio Issues with Incredible PBX
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.
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"
Originally published: Monday, January 17, 2022 Updated: 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.
Introducing OpenSIPS 3 for Incredible PBX and Debian 10
Today we’re pleased to introduce an updated OpenSIPS installer for Debian 10 featuring the latest release of OpenSIPS. Our previous tutorial with Debian 8 is now obsolete, an all-too-frequent occurrence in the open source world. Today’s open source SIP server lets you connect users to make and receive free as well as commercial calls worldwide. There’s excellent documentation making it easy to integrate into our existing Incredible PBX platform without hiring a consultant. It’s also straight-forward to secure without providing free phone service to every bad guy on the planet.
OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS Platforms, Call Centers, and many others. Source: opensips.org
We’ve often complained that the problem with many open source projects is that the developers get so focused on making money that they skimp on the documentation to encourage consulting work or participation in expensive conferences. We have found just the opposite with OpenSIPS. In fact, much of today’s implementation is based upon an excellent tutorial by the folks at PowerPBX. Down the road, if you find yourself in need of a consultant, their services would be a good place to start. What we’ve added to the PowerPBX design is security, support for clients behind NAT-based routers, and an integration scheme for Asterisk®, FreePBX®, and Incredible PBX® platforms so that you get the best of all worlds, a public facing SIP server with the UC feature set that most organizations expect. Last but not least, our turnkey GPLv2 installer will get you up and running in about 5 minutes.
Choosing an Appropriate Platform for OpenSIPS
Let’s begin by addressing the appropriate platform for an OpenSIPS server. The server needs to have a public IP address that is static, and the server should not be situated behind a NAT-based router. It only complicates things and is beyond the scope of what we plan to address. For those that are frequent visitors, you already know that we’ve been pushing everyone to kiss their local hardware goodbye and join the cloud revolution. When it comes to public-facing VoIP platforms like OpenSIPS, most of us don’t have a choice. You need a static IP address on the open Internet. And, for the sake of security, a KVM cloud platform is a must since older OpenVZ platforms don’t support the ipset component of IPtables which makes it easy to block hundreds of thousands of IP addresses without a performance hit on your server. Pure whitelist access simply isn’t an option if you wish to retain the functionality of a VoIP application such as OpenSIPS.
Ten to twenty gigabytes of disk space should be more than ample for OpenSIPS. The amount of RAM in your server depends upon the volume of calls your server will be handling. If it’s a dozen simultaneous calls then 1GB of RAM will suffice. If it’s 100,000 calls, then take a look at this article for tips on sizing your server. For today’s implementation, you’ll need a Debian 10 platform so a low-cost KVM provider including Digital Ocean, Vultr, and OVH should be fine.1
Choosing OpenSIPS Components to Deploy
We’ve divided up today’s tutorial into bite-sized pieces so that you can pick and choose where to stop implementing and start using. You do not need to have an Asterisk server to make and receive calls with OpenSIPS. However, OpenSIPS lacks voicemail and AutoAttendant/IVR components so, if those are a requirement, then you either need a VoIP service provider that offers them, or deploy a $50 Incredible PBX for the Raspberry Pi to add the missing pieces.
What OpenSIPS offers is a free server platform for worldwide SIP communications so that you, your friends, and business associates can call or connect from anywhere using freely available SIP softphones or any of dozens of SIP telephone instruments. We’ll stick with softphones for today, but hardware-based SIP telephones are equally simple to deploy.
This is not a criticism because it is one of the best tutorials we’ve ever used but, if you want to see how complex a typical OpenSIPS server deployment is, take a look at the PowerPBX tutorial we used as a starting point with OpenSIPS. We’ve compressed most of those procedures into a turnkey installer that only requires you to enter a MySQL root password of passw0rd (with a zero) once you have your Debian 10/64 platform up and running.
Deploying a Debian 10 Server Platform
Start by choosing a cloud provider that offers the 64-bit Debian 10 minimal platform as a deployment option. Most do. As noted, we recommend a KVM platform with support for ipset making it easy to block entire countries overrun with bad guys. Choose offerings with at least 1GB RAM and a 10GB drive to get started. Configure your Debian 10 server with a fully-qualified domain name (FQDN). This is critically important with our security design because we will assign all OpenSIPS users/extensions to this FQDN and reserve your server’s IP address purely for connections from service providers and Asterisk servers. This makes it all but impossible for anyone to hack into your server since most script kiddies launch attacks on IP addresses, not FQDNs. Using an unusual FQDN adds an extra layer of security, but that’s your call. If you lack the ability to assign FQDN aliases to a domain which you own, you can obtain a free FQDN from numerous sources including ChangeIP and point it to the IP address of your OpenSIPS server.
Installing OpenSIPS on a Debian 10 Server
Now the fun begins. Log into your Debian 8 server as root and issue the following commands to prepare for the OpenSIPS install:
cd /root wget http://incrediblepbx.com/opensips3.tar.gz tar zxvf opensips3.tar.gz rm -f opensips3.tar.gz
Make sure you have logged into your Debian 10 server as root using SSH or Putty from a desktop PC that you will use to manage OpenSIPS with a browser. The reason is because this IP address automatically will be whitelisted in the OpenSIPS firewall as part of the install process. Otherwise, you will need to manually log into SSH and whitelist the IP address of your desktop PC using /root/add-ip each time you wish to access the OpenSIPS Control Panel since TCP port 80 (HTTP) is not exposed to the public Internet as a security precaution.
To begin the install, issue this command: /root/install
As the install progresses, you’ll first be prompted to choose the GRUB install device. Press the spacebar on the first entry. Then press TAB and ENTER. When prompted for the SSH configuration, choose "keep local version" and then press TAB and ENTER. For the MariaDB setup, press ENTER when prompted for the current password. Type N when prompted whether to switch to unix_socket authorization. Then type Y to change the root password. Be sure to use passw0rd (with a zero) as your MySQL password, or the install will fail. This is NOT a security risk unless your Debian 10 root user account is compromised. And, in that case, it won’t matter anyway since the MySQL password could easily be changed. Type Y to remove anonymous users. Type Y to disallow remote root logins. Type Y to reload the MySQL privilege tables.
Next you’ll be prompted to set your timezone and TZ entries. For East Coast U.S., it’s 2,49,1,1 then America/New_York. Later you’ll be prompted twice for the MySQL root password. You must enter passw0rd (with a zero). When the OpenSIPS status screen displays, type Q to exit the display. There are a couple of steps where you will be prompted for input. Correct responses are indicated before the various prompts. Pay particular attention when you are prompted to change the SSH port from TCP 22 to a port number in the 1000-2020 range as a security precaution. We recommend using the year you were born because it will be easy for you to remember. When the install finishes and you log out of your server, the next SSH login will look like this where XXXX is the SSH port you chose and yyy.yyy.yyy.yyy is the OpenSIPS server address: ssh -p XXXX root@yyy.yyy.yyy.yyy
Although most of the configuration of your OpenSIPS server will be handled using a web browser and the OpenSIPS Control Panel GUI, we’ve included a few scripts in /root to assist with maintenance of your server platform. Here’s a brief summary of the script functions:
- pbxstatus – Status of your OpenSIPS server (image sample above)
- add-ip – Temporarily WhiteList IP address until next iptables-restart
- ban-ip – Permanently Ban an IP address
- unban-ip – Unban a previously banned IP address
- log-purge – Zero out all of the major Linux log files
- opensips-check – Assures OpenSIPS and RTPproxy are running (runs automatically)
- Fail2Ban BlackLists –
iptables -nL | grep -A100000 "opensips (" - IPset BlackList (KVM/OVZ7 platforms only) –
ipset list | sort
We secure your server in several ways: (1) by disguising the SSH port, (2) by locking down almost every port on your server with the IPtables firewall with the exception of the SIP ports, (3) by deploying Fail2Ban to scan your OpenSIPS log for errors and lock out attackers for an extended period of time, and (4) by deploying the IPset blacklist for KVM platforms. With this design, there is a symbiotic relationship between IPtables, Fail2Ban, and IPset. Therefore, it is critically important that you only restart these services using the iptables-restart command. NEVER issue other IPtables commands to restart or save your firewall settings.
Activating a SIP Server with OpenSIPS Control Panel
We don’t want to overload you on the first day with your new OpenSIPS 3 platform so we’ll walk you through the preliminary setup steps to create your SIP Domain. Then we’ll show you how to set up user accounts (also known as extensions). Finally we’ll walk you through setting up a trunk to make and receive calls from a commercial SIP provider. When we’re finished today, you’ll be able to make and receive calls using SIP URIs or DIDs which you have purchased from a provider. Then next week we’ll focus on integration of OpenSIPS with an Asterisk platform of your choice using Incredible PBX as an example. Once we’re finished, you’ll be able to handle user account registrations exclusively on your OpenSIPS server while leaving your Asterisk platform completely hidden from public exposure.
Logging into the OpenSIPS Control Panel
As deployed, the OpenSIPS Control Panel is accessible via web browser. As noted previously, HTTP Port 80 access is blocked by default unless the IP address of your desktop PC has been whitelisted either as part of the initial install or using the add-ip script in /root. Once your desktop PC’s IP address is whitelisted, point your browser to http://xxx.xxx.xxx.xxx/cp
The default Username is admin, and the default password is opensips. Once you’re logged in, immediately click on the Users icon in the upper-right corner of the dashboard. Then click the Edit Info pencil icon for user Admin and change your password. Click Save when done.
Creating Domains with OpenSIPS Control Panel
In the Left column of the Dashboard, you’ll see two tabs: Users and System. Click on the System tab to expose the available choices. Then choose the Domains option.
Domains are the essential building blocks in OpenSIPS. You can manage one or a hundred domains on a single OpenSIPS server, and each domain can have its own set of Users, Trunks/Gateways, and Dialplan rules. We’re actually going to create two domains, one for the IP Address of your OpenSIPS server and a second one for the FQDN of your OpenSIPS server. For added security, we will create all User accounts under the FQDN Domain. And we’ll reserve the IP Address Domain for DID Trunks/Gateways from registered, commercial SIP providers. This design allows attackers to attempt to register to accounts on your IP Address Domain until the cows come home, and they will never be successful because there are no existing SIP user accounts there. Keep it that way! With our OpenSIPS design, Fail2Ban will block attackers after a single failed registration attempt. And OpenSIPS itself will identify and block all SIP flood attacks using either Fail2Ban or IPset.
Now that you understand the design, let’s set up your domains. After choosing System -> Domains, enter the IP Address of your OpenSIPS server at the SIP Domain prompt. Then click Add New Domain followed by Reload on Server. Repeat the same steps to enter the fully-qualified domain name (FQDN) of your OpenSIPS server. When finished, you should see:
Creating Users with OpenSIPS Control Panel
We’ve already explained the security implications and reason for creating User accounts with your FQDN Domain only. Click on Users -> User Management -> Add New to get started. You can use Numbers (what we call Extensions in Asterisk) or Names. Our preference is to use Numbers for the User accounts and then to create Alias Names (as desired) for each User account. You can’t dial names from most SIP telephones. This also keeps the design similar to what many are used to in the Asterisk environment. A completed dialog would look something like the following. Use the Domain pull-down to choose your FQDN. Obviously, the passwords must be secure and must match. Then the Register button will be enabled to save. The actual Numbers used for Usernames are completely up to you.
Create at least a couple User accounts so that you can set up two SIP phones to call yourself and verify that everything is working. These User accounts become an integral part of the SIP URI to receive calls from any SIP phone in the world:
7701@opensips.yourdomain.com
Before you can actually answer an incoming call to your SIP URI, you’ll need to register the User account using either a softphone or SIP phone. We’ll do that next. But, first, let’s create an Alias to 7701 User so that folks can reach you by calling joe@opensips.yourdomain.com
Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the example below. Make sure that you select your FQDN Domain using the pull-downs for BOTH the Domain and Alias Domain fields. Then click Add to save.
Registering a Softphone to an OpenSIPS User Account
There are literally dozens of free SIP soft phones from which to choose. We covered some of our favorites for every platform in previous articles. For our purposes today, we recommend you choose one of the Linphone softphones which are available for the PC, Mac, Linux, Android, and iOS platforms. We also recommend signing up for a free Linphone.org SIP account which doesn’t cost you anything. For today, we will be configuring the softphone to register to your new OpenSIPS server.
Once you have downloaded and installed the Linphone client, go into the Preferences menu and make the following changes. Some depend upon your calling platform.
- Audio Codecs: PCMU, G722, PCMA
- Video Codecs: VP8, H264
- Call Encryption: None
- DTMF: RFC2833 only
- Send InBand DTMF: OFF
- Send SIP INFO DTMF: OFF
- SIP UDP 5060: Enabled
- SIP TCP 5060: Enabled
- Allow IPv6: Disabled
Then set up a new SIP Proxy account: Username (7701), Password (as defined), Domain: your FQDN not IP address, Transport: UDP, Outbound Proxy: OFF, Stun Server: stun.linphone.org, ICE: ON, AVPF: OFF, Push Notification: ON, Country Code Prefix: 1 (if required by your commercial SIP provider), Register: YES, Account Enabled: YES. HINT: You can call Alias Names via SIP URI, but you can only register to a SIP account using its actual Username.
Avoiding Lockouts with NeoRouter VPN
By design, Fail2Ban is unforgiving when it comes to failed registrations. A single failed registration will get an IP address banned for a full week. The reason is because the new bad guy strategy is to hit your server once to determine whether anybody is home. Then the creep bombards you later with an endless stream of registration attempts. With our design, nobody will be home when they return. The bad news is a single failed registration attempt by you or your users will also trigger a ban. There are several workarounds. The easiest is to set up the NeoRouter client on each of your machines including your OpenSIPS server and use the 10.0.0.x private network for access. These IP addresses never get banned. Our previous tutorial will walk you through setting up a free NeoRouter server and installing the free NeoRouter clients on your machines. The client software already is installed and running on your OpenSIPS server. It only requires that you log in using nrclientcmd and register to your NeoRouter server to obtain a private IP address. The other option is to install OpenVPN. Our previous tutorial will walk you through that process. The advantage of OpenVPN is that it’s supported directly on many SIP telephone instruments. The 10.8.0.x addresses are already whitelisted by our OpenSIPS installer.
There are other options to unban an IP address which has accidentally been snagged. First, almost all of the cloud providers include a Console option in their web portals. Second, you can log into your server via SSH from any non-blacklisted IP address to remove the banned IP address. Once you’re logged in, simply run this command using the IP address you wish to unban: /root/unban-ip xxx.xxx.xxx.xxx
Choosing Commercial SIP Providers
Recall that you cannot register to a SIP alias on your OpenSIPS server. We’ll take advantage of this restriction in setting up incoming calls from commercial providers’ DIDs. To set up Trunks from commercial providers so that you can not only receive incoming calls but also make outbound calls over their PSTN network connections, you must use providers that support IP address authentication rather than a SIP registration. Many providers support this including our platinum sponsor, Skyetel, as well as providers such as VoIP.ms, Anveo Direct, V1VoIP, and many others. In our OpenSIPS design, you also can use DIDs from providers that support SIP URI forwarding such as CallCentric and LocalPhone; however, you are limited to receiving inbound calls only. VoIP communications really shines here because you don’t have to choose a single provider to meet all of your communications requirements.
Skyetel is by far the easiest provider to set up with OpenSIPS. See our earlier tutorial for a special offer that will get you half-price calling for up to $500. Effective 10/1/2023, $25/month minimum spend required. Once you’re registered on the Skyetel site, add a new EndPoint Group using the IP address of your OpenSIP server and designate UDP 5060 as the access port. Sign up for a DID and map it to the OpenSIPS Endpoint Group. Done. In the OpenSIPS Control Panel, navigate to System -> Dynamic Routing and click Add Gateway. Using the template below, create 5 Proxy gateways for the following Skyetel data centers:
- skyetel-NW 52.41.52.34
- skyetel-SW 52.8.201.128
- skyetel-NE 52.60.138.31
- skyetel-SE 50.17.48.216
- skyetel-EU 35.156.192.164
Begin by whitelisting the IP addresses of your SIP providers in /etc/iptables/rules.v4 just below the existing 10.8.0.0/24 rule. The entries should look like this:
-I INPUT -s 52.41.52.34 -j ACCEPT
Once you’ve entered IP addresses for your providers, issue the command: iptables-restart
Next, we need to create what Asterisk users know as an Outbound Route. This tells OpenSIPS to send dialed numbers in 11-digit format to Skyetel for termination. We’ve already created the Dial Plan rule for calling out by dialing 1 plus a 10-digit number. So, while you’re still in the Dynamic Routing section of the OpenSIPS Control Panel, click on the Rules tab at the top of the template. Then click Add Rule. Begin by clicking Add ID button and choosing Group ID 0. In the Prefix field, type 1. Now click the Add GW button 3 times after choosing the Skyetel gateways in the following order from the GW pull-down list: skyetel-nw, skyetel-sw, and skyetel-se. Those are the three currently operational Skyetel gateways. When you’re finished, your template should look like the following. Then click the Add button to save the new rule. Click Reload Server to load the new rule into OpenSIPS. Then repeat this procedure leaving the Prefix field blank so that you can make 10-digit calls as well.
Finally, we need to create what Asterisk users know as an Inbound Route. This tells OpenSIPS where to send incoming calls from our Skyetel DID. OpenSIPS handles inbound routes by defining a User Alias for the Username to which you want to route the incoming DID calls. Click on Users -> Alias Management -> Add New Alias to get started. Fill in the form using the following template and then click Add.
- Username: 7701 (the extension to which to route the incoming calls)
- Domain: opensips.xyz.com (the FQDN of your OpenSIPS server)
- Alias Username: 18435551212 (the 11-digit Skyetel DID)
- Alias Domain: 11.12.13.14 (the IP address of your OpenSIPS server)
- Alias Type: dbaliases
Introducing the VoIP Blacklist
We’ve always dreamed of an effective VoIP Blacklist, and many have tried. But the crowd-sourced VoIP Blacklist at voipbl.org is the real deal. Everybody can post entries (including the bad guys) and, magically, most of the illegitimate entries get sifted out before the next day’s list is released. The list gets populated every night while you sleep. Here are the steps to install the VoIP Blacklist with IPset:
apt update && apt install ipset iptables netfilter-persistent ipset-persistent iptables-persistent cd /usr/local/sbin wget http://incrediblepbx.com/voipbl-update chmod +x voipbl-update sed -i 's|fail2ban restart|fail2ban restart\n/usr/local/sbin/voipbl-update|' iptables-restart iptables-restart ipset list voipbl ipset list voipbl | wc -l
Then create a cron job in /etc/crontab to run /usr/local/sbin/voipbl-update every day to update the VoIP blacklist.
1 4 * * * root /usr/local/sbin/voipbl-update > /dev/null 2>&1
Congratulations! You now have a functioning OpenSIPS 3 server that can process incoming calls from SIP URIs as well as DIDs. And you can make SIP URI and 11-digit PSTN calls using your SIP softphone that’s registered to your OpenSIPS server. See you next week. Enjoy!
Continue Reading: Best of Both Worlds: Safely Marrying Asterisk to OpenSIPS
Originally published: Monday, October 4, 2021
Need help with Asterisk? Visit the VoIP-info Forum.
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