Home » Cloud Computing (Page 11)

Category Archives: Cloud Computing

The Most Versatile VoIP Provider: FREE PORTING

A Golden Newbie: Incredible PBX 13-13.10 for Ubuntu 18.04.2


If you’re as big a fan of Ubuntu as we are, then you’ll be pleased to know that the month-old update to Ubuntu 18.04 LTS is rock-solid. It takes a brave soul to name anything Bionic Beaver, but Ubuntu pulls it off and makes you want to meet one face-to-face, just not in a bar after midnight. Well, St. Paddy would be proud. Today’s new Incredible PBX® 13-13.10 release brings you everything you could want in a PBX, and the icing on the cake is Ubuntu 18.04.2. The only drawback to Ubuntu 18.04 is that none of our $1/month VPS cloud providers support the platform just yet. But have no fear, both Digital Ocean and Vultr already do.1

Introducing 2019 Edition of Incredible PBX

This is our third major release of our flagship Incredible PBX 13-13 platform. In addition to today’s release for Ubuntu 18.04.2, it’s also available for Raspbian 8 as well as CentOS 6 and 7. It features 70+ new FreePBX® GPL modules plus all the latest components for OSS Endpoint Manager making SIP phone deployment with Asterisk® 13 a breeze. There also are terrific new backup and restore utilities which make migration and restoration of Incredible PBX platforms a snap. Finally, we’ve incorporated Skyetel SIP trunking in the build. It literally makes configuration of outbound and incoming calling a one-minute process. On the Skyetel side, create an Endpoint Group pointing to the IP address of your PBX, order one or more DIDs and point them to the new EndPoint Group. Done. On the Incredible PBX side, add Inbound Routes specifying the 11-digit numbers of your Skyetel DIDs and point each of them to the desired destination for incoming calls. Done. Outbound calls are automatically configured to use your Skyetel account. Our complete Skyetel tutorial is available here and includes up to a $250 usage credit with Skyetel’s new BOGO deposit match.2 Effective 10/1/2023, $25/month minimum spend required.

Creating an Ubuntu 18.04.2 Platform

If you plan to install Incredible PBX 13-13.10 using a cloud provider that supports Ubuntu 18.04.2, then creation of the Ubuntu 18.04.2 platform is as simple as clicking on the 64-bit OS as part of the creation of your 1GB RAM virtual machine. If you plan to use your own hardware, then any modern desktop computer will suffice. Begin by downloading the Ubuntu 18.04.2 ISO from here. Then create a bootable USB stick or assign the ISO as the boot device on your virtual machine platform. Here are steps for Ubuntu install using the server console:

  • Preferred language: English
  • Keyboard: English (US)
  • Install Ubuntu
  • Network interface (eth or wlan) from DHCP
  • Proxy (leave blank)
  • Ubuntu mirror (accept default entry)
  • Partitioning: Use Entire Disk
  • Choose Disk for Install (accept default usually)
  • File System Setup (choose Done)
  • Confirm Disk Install (Continue)
  • Profile Setup (create a username and password)
  • Install OpenSSH server (press Space Bar then Done)
  • Featured Server Snaps (leave blank)
  • Reboot Now (when prompted)
  • Remove installation media
  • Login using username created above
  • sudo passwd root
  • exit
  • Login as root with new root password
  • userdel username (that you created above)
  • nano -w /etc/ssh/sshd_config
  • Add: PermitRootLogin yes
  • save file
  • exit
  • Login as root using SSH or Putty

CAUTION: Don’t make any "improvements" to Ubuntu 18.04.2 after the initial install, or the Incredible PBX install may fail. It is designed for a base platform only!

Installing Incredible PBX 13-13.10

If you haven’t already done so, log into your Ubuntu 18.04.2 server as root using SSH or Putty. It’s important to log in from a desktop computer that you will be using to make changes on your server since this IP address will be whitelisted in the firewall as part of the installation process. Do NOT use the server console to install Incredible PBX, or you may not be able to log in from your desktop computer thereafter.

Before we begin the install procedure, let’s determine whether a swap file exists on your platform. If not, you’ll need to create one below as one of the first steps after downloading the Incredible PBX installer. Issue this command to determine if you have swap space: free -h

Now let’s download and install Incredible PBX 13-13.10. There are two flavors: the base install with the 70+ FreePBX GPL modules that comprise the web-based GUI to manage your PBX and the Whole Enchilada which adds 30+ Asterisk applications to the base install to provide TTS support, voice recognition, news and weather TTS apps, AsteriDex, telephone reminders, and much more. Here are the steps. Be sure to uncomment the create-swapfile-DO entry if you are lacking a swapfile.

cd /root
wget http://incrediblepbx.com/incrediblepbx-13-13.10U-LEAN.tar.gz
tar zxvf incrediblepbx-13-13.10U-LEAN.tar.gz
rm -f incrediblepbx-13-13.10U-LEAN.tar.gz
#./create-swapfile-DO
./Incredible*

There are two phases to the base install. You’ve just completed Phase #1. After your server reboots, log back in and kick off the Incredible PBX installer a second time. Don’t disappear immediately. On some cloud platforms, you may be asked whether to preserve your existing SSH setup. Choose the Keep Local Version default. On all platforms, you’ll be prompted for two additional responses in the first few minutes. At the first prompt, simply press ENTER to continue. At the second prompt, enter the country code to associate with your PBX. For those in the United States, the code is 1. We assume others are more familiar with their country code than Americans are. 😉

cd /root
./Incredible*

Make some careful notes when the install finishes. Then press ENTER to reboot your server.

If you don’t plan to use the Incredible PBX applications, then your install is complete after the reboot. Each time you log in to your server, the Automatic Update Utility will run to provide late-breaking updates that may affect the security of your server. So make sure you log in to the Linux CLI at least once a week to stay safe!

Assuming you’ve already created a very secure root password (update it by running passwd), perform the following 5 Steps to get everything locked down:

  1. Create an admin password for GUI access: /root/admin-pw-change
  2. Create an admin password for Apache web access: htpasswd /etc/pbx/wwwpasswd admin
  3. Configure the correct timezone for your server: /root/timezone-setup
  4. Retrieve your PortKnocker setup like this: cat /root/knock.FAQ
  5. Add IPtables WhiteList entries for remote access: /root/add-ip or /root/add-fqdn

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the first step above. HINT: You can always change it if you happen to forget it.

To get a basic system set up so that you can make and receive calls, you’ll need to add a VoIP trunk, create one or more extensions, set up an inbound route to send incoming calls to an extension, and set up an outbound route to send calls placed from your extension to a VoIP trunk that connects to telephones in the real world. You’ll also need a SIP phone or softphone to use as an extension on your PBX.

Continue Reading: Configuring Extensions, Trunks & Routes

Installing Incredible PBX 13-13 Whole Enchilada

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

The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. To run it, issue the following commands:

cd /root
./Enchilada*

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

amportal a ma install avantfax
amportal a r



Installing Incredible Fax with HylaFax/AvantFax

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

cd /root
./incrediblefax13_ubuntu18.sh

After entering your email address to receive incoming faxes, you’ll be prompted several times to choose options as part of the install. Simply press the ENTER key at each prompt and accept all of the defaults. When the install finishes, make certain that you reboot your server to bring Incredible Fax on line. There will be a new AvantFax option in the Incredible PBX GUI. The default credentials for AvantFax GUI are admin:password. Be advised that there remain a couple of quirks on the Ubuntu 18.04 platform. First, after entering your credentials, you may get a timeout error with your browser. Simply press the Reload/Refresh icon in your browser, and the default AvantFax menu will appear. Second, you will need to set your email delivery address and a new password for AvantFax manually. Click on the Settings option in the upper right corner of the dialog. When you save your settings, you may again experience a timeout event. Click the Reload/Refresh button on your browser again, and AvantFax will come back to life.

NAT-Based Router and Dynamic IP Wrinkles

If your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060 traffic to the private IP address of your PBX. While this isn’t technically necessary to complete calls with registered trunk providers, there are others such as Skyetel that don’t use SIP registrations where failure to redirect UDP 5060 would cause inbound calls to fail.

The Incredible PBX GUI is accessed using a web browser pointed to the IP address of your server. As part of the password setup, you created an admin password for the Incredible PBX GUI, a.k.a. FreePBX GUI. Login now using your favorite browser. If you have forgotten your admin password, you can reset it by logging into your server as root using SSH: /root/admin-pw-change. Once you’ve logged into the GUI, your first task is to record the public and private IP addresses for your server. This eliminates 99% of the problems with one-way audio on calls where your server is sitting behind a NAT-based router. Navigate to Settings -> SIP Settings and click on Detect Network Settings in the NAT Settings section of the template. Verify that the entries shown are correct and then click Submit followed by Apply Config.

Many Internet service providers assign dynamic IP addresses to customers. This poses issues with a PBX because SIP phones positioned outside your LAN need to be able to connect to the PBX. It also complicates SIP routing which needs both the public IP address and the private IP address of the PBX in order to route calls properly. In the previous section, you configured your PBX with these two IP addresses. The problem, of course, is that this public IP address may change when your ISP assigns dynamic IP addresses. Luckily, many ISPs rarely update dynamic IP addresses of their customers. For example, our home network has had the same dynamic IP address for more than four years. If this is your situation, then you have little to worry about. If the IP address ever changes, you can simply repeat the steps in the previous section. However, if your ISP regularly changes your public IP address, then you need an automatic way to keep your PBX configured properly. Otherwise you will start experiencing calls with one-way audio or no audio, and remote phones will no longer be able to connect to the PBX. We’ve developed a script to update the public IP address of your PBX. Depending upon your situation, all you need to do is run it hourly or daily to keep your PBX configured properly. To begin, first download the updater script after logging into your server as root:

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

Try running the script once to make sure it correctly identifies the public IP address of your server: /root/update-externip. Then add an entry to the end of /etc/crontab that schedules the script to run at 12:30 a.m. each night:

30 0 * * * root /root/update-externip > /dev/null 2>&1

Configuring Trunks with Incredible PBX

Before you can actually make and receive calls, you’ll need to add one or more VoIP trunks with providers, create extensions for your phones, and add inbound and outbound routes that link your extensions to your trunks. Here’s how a PBX works. Phones connect to extensions. Extensions connect to outbound routes that direct calls to specific trunks, a.k.a. commercial providers that complete your outbound calls to any phone in the world. Coming the other way, incoming calls are directed to your phone number, otherwise known as a DID. DIDs are assigned by providers. Some require trunk registration using credentials handed out by these providers. Others including Skyetel use the IP address of your PBX to make connections. Incoming calls are routed to your DIDs which use inbound routes telling the PBX how to direct the calls internally. A call could go to an extension to ring a phone, or it could go to a group of extensions known as a ring group to ring a group of phones. It could also go to a conference that joins multiple people into a single call. Finally, it could be routed to an IVR or AutoAttendant providing a list of options from which callers could choose by pressing various keys on their phone.

We’ve done most of the prep work for you with Incredible PBX. We’ve set up an Extension to which you can connect a SIP phone or softphone. We’ve set up an Inbound Route that, by default, sends all incoming calls from registered trunks to a Demo IVR. And we’ve built dozens of trunks for some of the best providers in the business. Sign up with the ones you prefer, plug in your credentials, and you’re done. The next section of this tutorial will show you the easier way, using Skyetel.

Unlike traditional telephone service, you need not and probably should not put all your eggs in one basket when it comes to telephone providers. In order to connect to Plain Old Telephones, you still need at least one provider. But there is nothing wrong with having several. And a provider that handles an outbound call (termination) need not be the same one that handles an incoming call (origination) and provides your phone number (DID). Keep in mind that you only pay for the calls you make with each provider so you have little to lose by choosing several. The PIAF Forum also has dozens of recommendations on VoIP providers.

With the preconfigured trunks in Incredible PBX, all you need are your credentials for each provider and the domain name of their server. Log into Incredible PBX GUI Administration as admin using a browser. From the System Status menu, click Connectivity -> Trunks. Click on each provider you have chosen and fill in your credentials including the host entry. Be sure to uncheck the Disable Trunk checkbox! Fill in the appropriate information for the Register String. Save your settings by clicking Submit Changes. Then click the red Apply Config button.

Using Skyetel with Incredible PBX

On the Raspberry Pi platform, all of the Skyetel trunks are preconfigured. All you need to do is sign up for Skyetel service in March to take advantage of the $50 Nerd Vittles special offer. 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 a $50 credit for your account by referencing the Nerd Vittles special offer. Greed will get you nowhere. Credit is limited to one per person/company/address/location. You can also take advantage of a 10% discount on your current service. Just open another ticket and attach a copy of your last month’s bill. See footnote 3 for the fine print.3 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch.

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

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

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

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

Configuring a Skyetel Inbound Route

Because there is no SIP registration with Skyetel, incoming calls to Skyetel trunks will NOT be sent to the Default Inbound Route configured on your PBX because FreePBX treats the calls as blocked anonymous calls without an Inbound Route pointing to the 11-digit number of each Skyetel DID. From the GUI, choose Connectivity -> Inbound Routes. You will note that we already have configured a Skyetel template for you. Simply edit the existing entry and plug in the 11-digit phone number (beginning with a 1) of your Skyetel DID . Set the Destination for the incoming DID as desired and click Submit. It defaults to extension 701.

If your PBX is sitting behind a NAT-based router, you’ll need to redirect incoming UDP 5060 traffic to the private IP address of your PBX. Then place a test call to each of your DIDs after configuring the Inbound Routes.

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

Configuring a Skyetel Outbound Route

If Skyetel will be your primary provider, it is preconfigured by default on the Raspberry Pi platform so you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. If you prefer another setup, choose Connectivity -> Outbound Routes.

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

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

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

Audio Issues with Skyetel

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

externip=xxx.xxx.xxx.xxx

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

Receiving SMS Messages Through Skyetel

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



Sending SMS Messages Through Skyetel

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

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

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

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

Configuring a Softphone for Incredible PBX

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

We recommend YateClient 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’ll need the IP address of your server plus your extension 701 password. Choose Applications _> Extensions -> 701 and write down your SIP/IAX Password. You can also reset it by running /root/update-passwords. Fill in the blanks using the IP address of your Server, 701 for your Username, and whatever Password you 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
*61 - Time of Day
TODAY - Today in History

If you are a Mac user, another great no-frills softphone is Telephone. Just download and install it from the Mac App Store.

Upgrading to IBM Speech Engines

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.

Getting Started with IBM Watson TTS Service

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

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

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

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

Getting Started with IBM Watson STT Service

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

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

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

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

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

Now try out the Incredible PBX Voice Dialer with AsteriDex by dialing 411 and saying "Delta Airlines."

Transcribing Voicemails with IBM Watson STT Service

We’ve included the necessary script to transcribe your incoming voicemails using IBM’s STT service. Navigate to the /usr/local/sbin folder and edit sendmailmp3.ibm. Insert your APIKEY in the password field and save the file. Now copy the file to sendmailmp3 and make the file executable: chmod +x sendmailmp3.

Using Gmail as a SmartHost for SendMail

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

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

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

Now issue the following commands. In the last step, press ENTER to accept all of the default prompts:

chmod 600 client-info
makemap -r hash client-info.db < client-info
cd ..
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/Makefile
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/databases
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc.gmail
sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf.errors
make
sendmailconfig

Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!

/etc/init.d/sendmail stop
/etc/init.d/sendmail start
apt-get install mailutils -y
echo "test" | mail -s testmessage yourname@yourdomain.com

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

Configuring a SIP URI Address for Your PBX

Setting up a SIP URI is a simple way to let anyone with a SIP phone call you from anywhere in the world and talk for as long and as often as you like FOR FREE. The drawback of SIP URIs is typically the security risk accompanying the SIP exposure you must provide to receive these calls. Here's the safe way using what we call a hybrid SIP URI. It works like this. Sign up for a VoIP.ms account and create a subaccount that you will register using the VoIPms trunk included in Incredible PBX. As part of the setup in the VoIP.ms portal, assign an Internal Extension Number to your subaccount, e.g. 789123. Make it random so you don't get surprise calls from anonymous sources. The extension can be up to 10 digits long. Next, sign up for a free iNUM DID, e.g. 883510009901234, in your VoIP.ms account. Using Manage DIDs in the portal, link the iNUM DID to your subaccount and assign one of the VoIP.ms POP locations for incoming calls, e.g. atlanta.voip.ms. Next, write down your VoIP.ms account number, e.g. 12345. Once you've completed these three steps and registered the VoIP.ms subaccount on your PBX, you now have two SIP URIs that are protected by your VoIP.ms credentials and don't require you to expose your SIP port to the outside world at all. These SIP URIs can be pointed to different destinations by setting up Inbound Routes using your VoIP.ms account number as one DID and setting up your iNUM number as the second DID. To reach your PBX via SIP URI, callers can use 12345789123@atlanta.voip.ms to reach the DID you set up for your VoIP.ms subaccount where 12345 is your VoIP.ms account number and 789123 is the Internal Extension Number for your subaccount. Or callers can use 8835100099012234@inum.net to reach the DID you set up using your iNUM number that was assigned by VoIP.ms. Don't forget to whitelist the VoIP.ms POP's FQDN for SIP UDP access to your PBX:

/root/add-fqdn voipms atlanta.voip.ms

If you wish to make SIP URI calls yourself, the easiest way is to first set up a free LinPhone SIP Account. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we'll get you started with one of our favorite (free) softphones, YateClient. It's available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for your LinPhone account. You’ll need LinPhone's FQDN (sip.linphone.org) plus your LinPhone account name and password. Fill in the Yate Client template and click OK to save your entries. Once the Yate softphone shows that it is registered, try a test call to one of our demo SIP URIs: sip:weather@demo.nerdvittles.com or sip:news@demo.nerdvittles.com.

Adding the NeoRouter Virtual Private Network

We've made it easy to set up a virtual private network between your PBX and your other computers. NeoRouter is a free VPN for up to 256 machines. It requires that you first set up a server for NeoRouter using a static IP address and preferably a fully-qualified domain name. This is covered in this Nerd Vittles tutorial. Once you have your NeoRouter server operational, adding your PBX to the VPN is easy. Simply run nrclientcmd and enter the FQDN of your VPN server together with your credentials. All clients on the VPN have an encrypted tunnel with private LAN addresses in the 10.0.0.x range. HINT: Setting up a NeoRouter VPN provides an easy way to get back into your server if the firewall ever locks you out since the 10.0.0.x subnet is automatically whitelisted as part of the initial install.

Using PortKnocker to Regain Access to Your PBX

And speaking of getting locked out of your server because you've forgotten to whitelist the IP address of your computer, there's another easy way to regain access: PortKnocker. The way the service works is you send sequential pings to 3 randomized TCP ports that are known only by you. They are listed in /etc/knock.FAQ. When your server detects a match, it will whitelist your new IP address allowing you to login using SSH or Putty. There also are PortKnocker utilities for both iOS and Android devices. Complete implementation details are available in this Nerd Vittles tutorial. If your PBX is sitting behind a router or firewall, don't forget to forward the three TCP ports from your router to the private LAN address of your PBX.

Planning Ahead for That Rainy Day

If you haven't already learned the hard way, let us save you from a future shock. Hardware fails. All of it. So spend an extra hour now so that you'll be prepared when (not if) disaster strikes. First, once you have your new PBX configured the way you plan to use it, make a backup of your PBX by running the Incredible Backup script: /root/incrediblebackup13

Copy down the name of the backup file that was created. You'll need it in a few minutes.

Second, build yourself a VirtualBox platform on your desktop PC using the Ubuntu ISO you previously downloaded. Once you complete the identical Incredible PBX install plus the Whole Enchilada upgrade and Incredible Fax (if used on your primary server), fire up the virtual machine and login as root with password as your password.

Next, create a /backup folder on your new VirtualBox PBX and copy the backup file from your main server to your VirtualBox server and restore it while logged into the VirtualBox PBX as root:

mkdir /backup
scp root@main-pbx-ip-address:/backup/backup-file-name.tar.gz /backup/.
/root/incrediblerestore13 /backup/backup-file-name.tar.gz

Verify that everything looks right by using a browser to access and review the settings in your new VirtualBox PBX. At a minimum, verify extensions, trunks, and routes.

Last but not least, if you're running Incredible PBX in the Cloud on Digital Ocean or Vultr, you can set up automatic backups of your server for only an extra dollar a month. It's the cheapest insurance your can buy. Enjoy!

Continue Reading: Configuring Extensions, Trunks & Routes

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

Originally published: Monday, March 18, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. With some providers including Digital Ocean and Vultr, Nerd Vittles receives referral credits when you sign up for service. This assists in keeping the Nerd Vittles lights burning brightly. So... thank you. []
  2. Skyetel is a Platinum Sponsor of Nerd Vittles and open source projects of Ward Mundy & Associates, LLC. []
  3. In the unlikely event that Skyetel cannot provide a 10% reduction in your current origination rate and/or DID costs, Skyetel will give you an additional $50 credit to use with the Skyetel service. []

Yowza! A Fault-Tolerant Incredible PBX Platform for $1/Mo.


Not that you needed another reason to sign up for Skyetel’s triple-redundant SIP hosting platform, but we have a new one for you anyway. A fault-tolerant VoIP platform always has been the Holy Grail for many organizations. When your business depends on telephone calls, the very last thing you want to fail is your phone system. When most administrators think of fault-tolerant, high availability (HA) servers, the first word that comes to mind is expensive. For example, FreePBX® charges $1500/node just for HA software licenses. Using Skyetel trunks with Incredible PBX® 13-13.10, you now can deploy a fault-tolerant solution including the cloud-based secondary server for about a dollar a month. And, unlike the FreePBX design, the primary and secondary servers need not be colocated. Today we’ll get you up and running.

Many SIP providers require users to register their PBX with their service before you can make and receive calls. Skyetel does not.1 Instead, Skyetel relies upon the IP address of the PBX to determine whether to relay calls to and from your PBX. Dynamic IPs are supported. This is incredibly useful in designing a fault-tolerant PBX platform. If SIP registration is required with your SIP provider, then it obviously takes considerably more engineering effort to enable and disable the redundant server platform since you can’t be registered to two different servers using the same accounts at the same time. With Skyetel, that’s never a problem. Let’s quickly review the Skyetel setup process, and you’ll see why their design facilitates fault tolerance without spending a fortune on additional hardware and software. Effective 10/1/2023, $25/month minimum spend required.

Once your Skyetel account is set up, there are basically two steps to enable connections with your PBX. First, you create an Endpoint Group with the IP address of your PBX. Second, you purchase one or more DIDs and point each of them to the Endpoint Group you just created. With Incredible PBX platforms, the Skyetel trunk and firewall setup already is in place. All you need to do is create an Inbound Route for each of your Skyetel DIDs using its 11-digit phone number. Ten and eleven-digit outbound calls, by default, always are routed to Skyetel for processing. So how do we handle fault-tolerance on the Skyetel side? It’s easy. Just add the IP address of the secondary PBX to the Endpoint Group that you set up for your primary PBX. That’s it. When Skyetel can’t send inbound calls to your primary PBX, it now will send them to your secondary server.


Now let’s address what has to happen at your end to bring up a fault-tolerant platform. We will assume that you already have set up and configured extensions, trunks, routes, and all desired bells and whistles on an Incredible PBX 13-13.10 platform somewhere. It really doesn’t matter where. We will also assume you have deployed SIP telephones for all of your users and registered them with your PBX. There are five simple steps to implement a fault-tolerant PBX architecture. First, you need a cloud-based Incredible PBX 13-13.10 server with static IP address and nightly-refreshed mirror image of your primary server. Second, SSH needs to be configured for SSH key-based authentication between the two servers. We’ll use rsync and SSH on the primary server to keep voicemails synchronized between your primary and secondary servers during the day. Third, we’ll add the IP address of the secondary PBX as the secondary registration IP address on each of your SIP phones. Fourth, you’ll need to copy over the firewall and whitelist rules from your primary server to your second server and restart IPtables after making certain that you’ve whitelisted the IP addresses of BOTH servers on the primary server. Fifth, we’ll add the IP address of the secondary PBX to the Endpoint Group of your primary PBX on the Skyetel site. That’s it. Once you’ve completed these five steps, your users will never miss a call. Since Skyetel doesn’t sit in the middle of the RTP stream on calls that are underway, callers may never even know their primary PBX failed.

So let’s address the worst case scenario. Your primary server has an unrecoverable system failure in the middle of the work day, and you have no backup. What do you lose? Not phone calls! And, assuming you’re making hourly dumps of your voicemails to the secondary PBX, you only would lose whatever voicemails arrived during the minutes preceding the system failure. If an hour is too painful in your operation, then make voicemail dumps more frequently. The only other data that would be lost is the CDR data since the last nightly backup was restored. If that’s too painful, that data also can be archived on a more frequent basis using a simple bash script. And, just to repeat, the total additional cost for this fault-tolerant platform is approximately $1 a month depending upon the VPS provider you choose, and we have a half dozen that we recommend AND USE at that price point. Let’s get started.

Building a Secondary Redundant Server

Start by signing up for a cloud-based OpenVZ VPS platform with one of our recommended providers. Server locations and special signup details are documented in our article. Average cost is about $1/month on an annual contract with 1Gbit port or *free 1Gbit port upgrade on request based upon LowEndBox offer. Protect yourself by paying with PayPal which gives you 6 months to dispute a charge if the provider happens to go belly up. NOTE: Performance is almost directly proportional to annual cost with our Tier 1 and Tier 2 providers.

ProviderRAMDiskBandwidthPerformance as of 12/1/19Cost
CrownCloud KVM (LA)1GB20GB +
Snapshot
1TB/month598Mb/DN 281Mb/UP
2CPU Core
$25/year
Best Buy!
Naranjatech KVM (The Netherlands)1GB20GB1TB/monthHosting since 2005
VAT: EU res.
20€/year w/code:
SBF2019
BudgetNode KVM (LA)1GB40GB RAID101TB/monthAlso available in U.K PM @Ishaq on LET before payment$24/year
FreeRangeCloud KVM (Ashburn VA, Winnipeg, Freemont CA)1GB20GB SSD3TB/monthPick EGG loc'n
Open ticket for last 5GB SSD
$30/year w/code:
LEBEGG30

If you can’t bear repeating this step in the event of a provider implosion, then consider one of the $5/month cloud platforms offered by Digital Ocean or Vultr.2 Ask yourself if it’s worth $4/month to eliminate the risk of having to spend an hour rebuilding your secondary PBX platform. Regardless of what you decide, here’s the drill. Set up a 64-bit CentOS 6 platform on your VPS. Since you already are familiar with the Incredible PBX 13-13 setup procedure, just follow last week’s quick-and-dirty checklist to set up your new secondary server. If your primary server also has the Whole Enchilada or Incredible Fax installed, we recommend you install the same components on your secondary server. If your secondary server is CentOS 7-based, issue the following command to assure that key-based authentication will work: chmod 711 /root. When you’re finished, write down the IP address of your secondary server.

Setting Up SSH Key-Based Authentication

In order to automate the process of loading updates from your primary server to your secondary one, we need a way to login between the servers without being prompted for a password. Key-based authentication provides that feature. Begin by whitelisting the IP addresses of BOTH your primary and secondary PBXs on BOTH of your servers. Using add-ip in /root, add server1 entry: ./addip server1 12.34.56.78. Then add server2 entry: ./addip server2 23.45.67.89. When prompted, choose the 0 option to whitelist all ports.

Next, store the IP addresses of server1 and server2 on your primary server so we can find them when we need them for future tasks:

echo 12.34.56.79 > /etc/pbx/server1
echo 23.45.67.89 > /etc/pbx/server2

Next, on your primary server, generate an SSH key-pair: ssh-keygen. Press Enter when prompted for the key’s file name. Press Enter when prompted (twice) for a passphrase.

Then copy the new key to your secondary server using the IP address of your secondary server. Enter the root password for your secondary server when prompted.

ssh-copy-id root@$(cat /etc/pbx/server2)

Now test logging into your secondary server using SSH from your primary server. You should not be prompted for a password. Close the secondary server SSH session by typing exit.

ssh root@$(cat /etc/pbx/server2)
exit

Restoring Backup Image to Secondary Server

We will automate these steps down the road, but let’s first test the procedure to make sure things work as expected. Begin by creating a /backup folder on both servers. Then install the latest backup and restore scripts on both of your servers:

mkdir /backup
cd /root
rm -f incrediblebackup
rm -f incrediblerestore
wget http://incrediblepbx.com/incrediblebackup13.tar.gz
tar zxvf incrediblebackup13.tar.gz
rm -f incrediblebackup13.tar.gz

On BOTH servers, verify that you have the latest backup and restore scripts. The following command should return the filenames of your backup and restore scripts:

grep PROCEEDNOW= /root/inc*

On BOTH servers, issue the following commands so that we can automate the backup and restore process without being prompted:

sed -i 's|PROCEEDNOW=false|PROCEEDNOW=true|' /root/incrediblebackup13
sed -i 's|PROCEEDNOW=false|PROCEEDNOW=true|' /root/incrediblerestore13

Finally, on both servers, make sure you have logged out and back in at least once so that the Automatic Update Utility has loaded the latest patches on both platforms.

Now let’s take a backup snapshot of your primary server: /root/incrediblebackup13

Copy the backup file to the secondary server at the server2 IP address below:

scp /backup/$(ls -t /backup | head -n 1) root@$(cat /etc/pbx/server2):/backup/.

Next, login to your secondary server as root using SSH. We want to restore the backup file that was just copied from the primary server. Keep in mind that our restore methodology overwrites files with the same names on the secondary server but it does not erase files on the secondary server which were not contained in the primary’s backup. With the exception of the /root directory, the latest iteration of Incredible Restore cleans out the other affected directories including voicemail files on the secondary server. This is especially important because voicemails may have been erased on the primary server since the last restore, and voicemail files get renumbered each time a voicemail is removed. Thus, failure to remove previous collections of voicemail files before restoring a new backup could produce a royal mess on the secondary server especially where the primary server processes numerous voicemails for multiple users every day. One other item of interest concerns primary servers behind NAT-based routers. Ordinarily, you would set the external IP address and local networks in SIP Settings to avoid NAT problems such as one-way audio. Since we may be copying a NAT-based backup to a public facing secondary PBX, we don’t want those settings carried over. We’ve addressed this by deleting the entries whenever you choose automated restores using the PROCEEDNOW=true flag. The full backup including these NAT settings is preserved if you execute a restore using the PROCEEDNOW=false flag. Finally, if you were using the backup and restore tools for normal server recovery, we obviously would want to include the /etc/crontab file in the restore. However, for fault tolerant deployments, we want to preserve separate cron jobs on the primary and secondary PBXs. We handle this using the PROCEEDNOW flag as described above. If you set it to true, we preserve /etc/crontab. Otherwise, we restore it from the backup file. Now restore the backup on the secondary PBX:

/root/incrediblerestore13 /backup/$(ls -t /backup | head -n 1) 

Although not recommended, we’ve been successful restoring a CentOS6-based primary server to a CentOS7-based secondary platform. Take a minute to actually examine the contents of the backup on the secondary server by logging into the FreePBX GUI with a browser. Be sure to check the extensions, trunks, and routes to verify that they match the entries on your primary server. If your primary server is behind a NAT-based router, check the SIP Settings to make certain the NAT entries for external IP address and localnets weren’t carried over to the public-facing secondary PBX.


Refreshing Spool Data to Secondary Server with rsync

Now we want to test updating voicemails and recordings from your primary server to the secondary. Using a SIP phone, connect to *701 and leave yourself a new voicemail. Verify that it exists in /var/spool/asterisk/voicemail/default/701/INBOX. Simulate the procedure that will be used periodically to copy new voicemail messages and recordings to your secondary server. Execute the following commands to verify the new voicemail was transferred:

rsync -a -e "ssh" /var/spool/* root@$(cat /etc/pbx/server2):/var/spool
ls -all /var/spool/asterisk/voicemail/default/701/INBOX

Adding the Secondary Server to Skyetel

As noted above, we need to add the IP address of your secondary server to the Endpoint Group of your primary server on Skyetel. After logging into your Skyetel account, choose Endpoints -> Endpoint Groups and click the gear icon beside your primary server’s endpoint group. Add an additional entry to the Group with a Priority of 2. Your entries should look similar to this:

Within a few minutes, the Endpoint Health dashboard should display three green dots for both of your servers signifying that all three Skyetel data centers can connect to both of your servers.

Adding Server2 Key-Based Authentication

On your secondary server, let’s set up key-based authentication to the primary PBX. Here are the commands to set it up and test it. You should get logged in and back out without a password prompt.

ssh-copy-id root@$(cat /etc/pbx/server1)
ssh root@$(cat /etc/pbx/server1)
exit

Choosing Compatible Phones for High Availability

Now that we have our HA platform in place, we need to configure your SIP phones to continue to work when the primary server fails and things switch over to the secondary PBX. For this to work, you’ll need SIP phones that are compatible with HA technology. Most are but some are not. Look through the registration menu on your SIP phone and enter the IP address of the secondary PBX in the field provided. Adjust the server timeout value to reflect the number of seconds your users can tolerate without a working phone. On the primary PBX, don’t forget to whitelist the public IP addresses associated with each of your SIP phones using add-ip! As for supported phones, here’s a hint. Most Digium, Snom, Yealink, Grandstream, Fanvil, and Aastra 6700i and 9000i series phones are safe bets. And here’s what a SIP extension setup would look like on Yealink’s popular T46G.


Reminder: Do NOT use a private IP address (as shown) for the secondary SIP registration, and only use a private IP address for the primary PBX if both the PBX and the SIP phone are behind the same NAT-based router.

Migrating Travelin’ Man 3 Firewall to Secondary PBX

You’ll find it much easier to manage one firewall and then copy those firewall settings from your primary server to your secondary PBX. There are 3 sets of files that need to be copied from the primary PBX before restarting the secondary’s firewall: (1) iptables and ip6tables from /etc/sysconfig, (2) all iptables* files in /usr/local/sbin, and (3) *.iptables files in /root. Make certain you have whitelisted the IP addresses of the primary and secondary PBXs using /root/add-ip before issuing these commands on the primary server:

cd /etc/pbx
scp /etc/sysconfig/iptables root@$(cat server2):/etc/sysconfig/.
scp /etc/sysconfig/ip6tables root@$(cat server2):/etc/sysconfig/.
scp /usr/local/sbin/iptables* root@$(cat server2):/usr/local/sbin/.
scp /root/*.iptables root@$(cat server2):/root/.
ssh root@$(cat server2) "service iptables restart; service fail2ban restart"
ssh root@$(cat server2) "iptables-custom"

Automating the Daily Synchronization Process

We’ll use three cron scripts to keep the data on your primary and secondary servers in sync. These scripts automatically manage the backup sets by removing all but the latest backup on both servers. Keep in mind that, if you add new packages to your primary server, you also will need to add the same packages to your secondary server. Aside from that caveat, everything else should remain synchronized using these three simple scripts. On the primary server, we’ll use backup1restore2 to make the nightly backup and copy it to the secondary server, and we’ll use rsync-spool-1to2 to execute the hourly spool sync to catch changes in voicemails, recordings, and fax data. On the secondary server, we’ll use restore2backup1 to restore the backup from the primary server each night. Let’s first put the cron scripts in place, and then we’ll set up the cron jobs.

On the primary server, issue the following commands to install the two scripts:

cd /root
wget http://incrediblepbx.com/backup1restore2
chmod +x backup1restore2
wget http://incrediblepbx.com/rsync-spool-1to2
chmod +x rsync-spool-1to2

On the secondary server, issue the following commands to install the restore script:

cd /root
wget http://incrediblepbx.com/restore2backup1
chmod +x restore2backup1

Before setting up the cron entries, you have a few decisions to make. First, decide how often you wish to run the backup/restore scripts. If your PBX configuration rarely changes, you may not need to run the backup and restore scripts every night. Second, decide what time to run the backup and restore scripts. The restore script on the secondary server should always be run about 30 minutes AFTER the backup script runs on the primary server. The reason should be obvious. Remember to take into consideration the time zones of your two servers when making the time calculations. Third, decide how frequently to run the spool synchronization script from the primary server. This will depend upon how busy your primary server is and how much data (voicemails) you’re willing to lose in the event of a catastrophic failure that occurs after the last sync operation. There’s probably little need to run the synchronization script while everyone is sleeping unless you get frequent fax deliveries overnight. Finally, the backup of the primary server requires that Asterisk be shut down for at most a few minutes so schedule the backups during a time when there is no server activity on your primary server. Your test backup that we ran above should provide some idea of how long the operation will take on your PBX. Once you have tackled these issues, edit /etc/crontab on both your primary and secondary servers. Add the cron jobs to the end of the files after verifying server time on BOTH servers using date.

NOTE: We strongly recommend setting the time zone on the secondary server to match the primary server’s time zone using the /root/timezone-setup script.

On the primary server, here is a sample entry to run the backup script as the root user once a day at 2:15 am (local time on primary server):

15 2 * * * root /root/backup1restore2 >/dev/null 2>&1

On the secondary server, this entry will run the restore script once a day at 2:45 am assuming secondary server is in the same time zone as primary server:

45 2 * * * root /root/restore2backup1 >/dev/null 2>&1

On the primary server, this entry runs rsync spool script every hour on the half hour:

30 * * * * root /root/rsync-spool-1to2 >/dev/null 2>&1

Testing Fault Tolerance on Your Platform

Once you’ve put all the pieces in place, the easiest way to test the HA functionality is to shut down Asterisk on the primary PBX: amportal stop. Also disable the rsync cron job in /etc/crontab. Depending upon the timeout seconds you configured on your SIP phones, you should be able to make calls through the secondary server shortly. Don’t forget to restart Asterisk on your primary PBX: amportal start. Note also that you will need to shut down Asterisk on the secondary server for a few minutes after restoring the primary server. This will force all of your SIP phones to re-register with the primary server. And, in a real-world outage of considerable duration, it may be necessary to do a reverse-rsync of the spool directories from the secondary server (using the template below as an example) to assure that no voicemails and other spool files were lost during the outage. Then it’s safe to once again enable the rsync cron job on the primary server.

amportal stop
rsync -a -e "ssh" /var/spool/* root@$(cat /etc/pbx/server1):/var/spool
sleep 900
amportal start

Originally published: Thursday, March 7, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. Skyetel is a platinum sponsor of Nerd Vittles and the open source projects of Ward Mundy & Associates, LLC. March is the final month to take advantage of the Nerd Vittles $50 usage credit when you sign up with Skyetel. []
  2. Digital Ocean and Vultr provide modest referral credits to Nerd Vittles for those that use our referral code. It in no way colors our recommendations regarding these two providers, both of whom we use extensively. []

Now Serving: The Incredible PBX 13-13 Whole Enchilada

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

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

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

Installing a Base CentOS Operating System

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

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

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

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

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

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

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

Installing Incredible PBX 13-13 LEAN

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

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

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

Using the Incredible PBX 13-13 Web GUI

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

Most of the configuration of your PBX will be performed using the web-based Incredible PBX GUI with its FreePBX 13 GPL modules. Use a browser pointed to the IP address of your server and choose Incredible PBX Admin. Log in as admin with the password you configured in the previous step. HINT: You can always change it if you happen to forget it.

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

Continue Reading: Configuring Extensions, Trunks & Routes

Upgrading to Incredible PBX Whole Enchilada

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

The Whole Enchilada upgrade script now is included in the Incredible PBX LEAN tarball. If you have an earlier release, you may need to download the Whole Enchilada tarball as documented below. Upgrading to the Whole Enchilada is simple. Log into your server as root and issue the following commands. Try issuing just the last command first to see if the enchilada upgrade script already is in place. Otherwise, execute all of the commands below. Be advised that the upgrade will overwrite all of your existing Incredible PBX setup including any extensions, trunks, and routes you may have created previously. You also will be prompted to reset all of your passwords as part of the upgrade.

cd /root
./Enchilada*

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

amportal a ma install avantfax
amportal a r

Installing Incredible Fax with HylaFax/AvantFax

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

cd /root
./incrediblefax13.sh

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

Upgrading to IBM Speech Engines

If you’ve endured Google’s Death by a Thousand Cuts with text-to-speech (TTS) and voice recognition (STT) over the years, then we don’t have to tell you what a welcome addition IBM’s new speech utilities are. We can’t say enough good things about the new IBM Watson TTS and STT offerings. With IBM’s services, you have a choice of free or commercial tiers. Let’s put the pieces in place so you’ll be ready to play with the Whole Enchilada.

Getting Started with IBM Watson TTS Service

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

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

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

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

Getting Started with IBM Watson STT Service

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

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

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

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

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

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

Adding Skyetel Trunks to Incredible PBX

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

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

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

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

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

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

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

Configuring a Skyetel Inbound Route

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

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

Configuring a Skyetel Outbound Route

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

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

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

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

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

Audio Issues with Skyetel

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

externip=xxx.xxx.xxx.xxx

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

Receiving SMS Messages Through Skyetel

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



Sending SMS Messages Through Skyetel

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

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

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

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

Using Gmail as a SmartHost for SendMail

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

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

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

Now issue the following commands:

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

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

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

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

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


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

Continue Reading: Configuring Extensions, Trunks & Routes

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

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


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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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

Adding SIP URI Dialing to Asterisk for Free Worldwide Calling



Since giving up on Google Voice, we’ve been extolling the virtues of SIP URI dialing which gives you unlimited free calls to anyone else in the world that happens to have their own SIP URI address. SIP URIs look very much like email addresses except they’re used to share phone conversations instead of email messages. And, as we’ve mentioned previously, if everyone in the world had their own SIP URI, paying for phone calls would become a thing of the past. We hope you’ll join us in making that happen. As a fallback, give our $50 credit at Skyetel a try.

One of the drawbacks of Asterisk® PBXs using the FreePBX® GUI has been the inability to place outbound SIP URI calls from SIP phones registered as extensions on the PBX. Today we first want to address that shortcoming. Our SIP URI dialing solution for Asterisk should work with any FreePBX-based implementation including Incredible PBX® and Issabel as well as on Raspberry Pi platforms. We’ll wrap things up by providing some tips on obtaining and deploying your own SIP URI at little or no cost and pointing you to some excellent resources that facilitate calling millions of SIP phones around the world at zero cost. All you need is an Internet connection, and we’ll point you to a terrific softphone to begin your adventure.

Let’s begin by examining why SIP URI dialing is a problem with FreePBX. The reason is pretty simple. FreePBX interprets dial strings by matching them against some rules to determine whether you’re making an internal call or a call outside your PBX. It matches internal calls against a list of available internal extensions. External calls are matched against rules defined in your outbound routes which are associated with trunks. Since SIP URI calls don’t match any extension or outbound route, the caller receives a congestion tone.

The traditional workaround has been to define a custom extension using the FreePBX GUI which points to a SIP URI. Then the user can dial the custom extension, and the call will be routed to the defined SIP URI. These custom extensions also can be defined in extensions_custom.conf within the from-internal-custom context. For example, the following dialplan code would let users dial 411 to reach AT&T’s Toll-Free Directory Assistance: exten => 411,1,18005551212@switch.starcompartners.com.

But there’s a better way. Wouldn’t it be nice to be able to dial any SIP URI from a softphone or to store SIP URI addresses in the phonebook of your SIP phone?1 Well, now you can. Before we actually put the dialplan code in place, let us explain how this will work. First, FreePBX still needs to be able to distinguish a SIP URI call from a "regular call." The reason this gets tricky is because Asterisk typically throws away the destination hostname when you place a call. For example, calls to 8005551212 and 8005551212@sip2sip.info are processed by Asterisk in exactly the same way, i.e. dropping the host address before dialing.

Using the new dialplan code in the next section, here’s how calls will be processed:

User dials                    Asterisk processes call as
------------------------      ---------------------------------------------
701                           internal call to local extension 701
4045551212                    external call using NXXNXXXXXX outbound route
2233435945@sip2sip.info       SIP URI call to Lenny by acct at sip2sip.info
lennybgood@sip2sip.info       SIP URI call to alias lennybgood@sip2sip.info 

Cautionary Notes: Our code should work fine with any Asterisk 13 and FreePBX 13 or Incredible PBX deployment on any Linux platform; however, with servers other than Incredible PBX, make sure you have added the following entries to sip_general_custom.conf, or you can configure them in the GUI by making the changes in Settings -> Asterisk SIP Settings -> Chan SIP Settings:

srvlookup=yes
allowguest=yes

You also need to test a traditional outbound call (e.g. 8005551212) immediately after you finish the install procedure. Monitor the Asterisk CLI (asterisk -rvvvvvvvvvv) and observe the first few lines of the log after you place a call. The second line will show SIPDOMAIN which should be either the FQDN of your server or an IP address depending upon how you registered your softphone extension. The first line should display the MyDomain variable. If it is empty or doesn’t match the SIPDOMAIN entry, the outbound call will fail. To fix it, add an entry to the Asterisk database from the Asterisk CLI using syntax like the following: database put MyDomain FQDN 10.0.0.11 or database put MyDomain FQDN sip.me.com where 10.0.0.11 or sip.me.com matches the SIPDOMAIN entry shown on the second line. Then retry your outbound call, and it should complete successfully. We’ve tested this back to the early Asterisk 11 days with FreePBX 2.11 without any problems. If your calls still fail, then you will probably need to remove the new code from your platform until you upgrade to a more current version of Asterisk and FreePBX. The code hasn’t been tested with FreePBX 14 and 15.

Finally, you may want to manually set the CallerID for your outgoing SIP URI calls. From the Asterisk CLI, issue a command for every extension from which you will be placing SIP URI calls, e.g. extension 701 syntax: database put 701 user_sipname "Nerd Uno"

Enabling SIP URI Dialing with FreePBX

To enable SIP URI dialing from phones registered with your Asterisk PBX, we’ll modify the dialplan in order to detect SIP URI dial strings entered into a softphone or retrieved from a phonebook associated with almost any SIP phone. When a SIP URI dial string is detected, we’ll send the call out as requested rather than passing the call through the outbound routes and trunks associated with your PBX. All of this dialplan code is open source and is licensed pursuant to the GPL2 license.

SECURITY ALERT: Never use the SIP URI MOD on a server with a publicly-exposed SIP port as it is possible for some nefarious individual to spoof your FQDN in the headers of a SIP packet and easily gain outbound calling access using your server’s trunk credentials.

FEB. 21 UPDATE: There was a bug in the original code which caused some internal calls to fail including calls to a DISA extension. Simply install the application again, and it will overwrite the previous version.

MAR. 5 UPDATE: A bug was discovered in previous releases that treated 911 and 933 calls as internal calls when, in fact, they should have been routed out using your outbound trunks. Simply install the application again, and it will overwrite the previous version.

MAR. 13 ALERT: This software is not compatible with the Debian, Raspbian, and Ubuntu platforms.

To begin or update your installation, log in to your PBX as root using SSH or Putty and issue these commands:

cd /tmp
wget http://incrediblepbx.com/sipuri-mod.tar.gz
tar zxvf sipuri-mod.tar.gz
rm -f sipuri-mod.tar.gz
./install-sip-uri-mod.sh

Obtaining Your Own SIP URI

There are a number of ways to obtain your own SIP URI. Perhaps the easiest is to set up the open Incredible PBX cloud platform that we introduced several weeks ago. Then you can create as many SIP URIs as you like, and they can be used to perform any task that’s available with Asterisk. If you’re not quite ready to make that leap, a free or almost free SIP URI is available from the following sources. VoIP.ms provides a SIP URI for every subaccount you create. Just set up an internal extension number for the subaccount, and that becomes a SIP URI to connect back to your registered server or SIP phone. In the alternative, VoIP.ms will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. CallCentric provides a SIP URI matching your account number which can be reached @in.callcentric.com. CallCentric will also provide you with a free iNUM DID which can be reached at the following IP address: 81.201.82.50. LocalPhone provides the same two options as CallCentric: you can be reached by your account number @localphone.com. Or the LocalPhone-assigned iNUM DID can be reached @81.201.82.50. Then there’s pbxes.org. Your account name can be used for SIP URI access @pbxes.org. And, of course, if you’re a 3CX user, you can set up a SIP URI for each extension on your PBX. Just navigate to the Options tab of the desired extension(s) and enter a unique SIP ID for each extension. The SIP URI becomes SIPID@YOUR-3CX-FQDN. SIP URI calls to 3CX Clients on smartphones are also free! This list is not exhaustive. There are now more than 2,000 VoIP networks that support SIP URI access. Using a SIP URI dialing prefix, call any of the referenced networks @sipbbroker.com.2

Choosing a SIP Phone or Softphone

You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we’ll get you started with one of our favorite (free) softphones, YateClient. It’s available for almost all desktop platforms. Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for an extension on your PBX. You’ll need the IP address of your server plus your extension number and its password. Fill in the Yate Client template using the IP address of your PBX as well as your extension credentials. Click OK to save your entries.

Once the Yate softphone shows that it is registered, try a test call to Lenny using one of the following SIP URIs: 2233435945@sip2sip.info or 883510001198938@81.201.82.50. Better yet, try out a few Incredible PBX samples from the public server we previously deployed:

Yahoo News Headlines    - news@demo.nerdvittles.com
Weather by Zip Code     - weather@demo.nerdvittles.com
Directory Assistance    - information@demo.nerdvittles.com
Lenny for Telemarketers - lenny@demo.nerdvittles.com

Originally published: Monday, February 11, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. Special thanks to Olivier Adler and voip-info.org for their early work on SIP URI dialing with Asterisk. []
  2. Some of our links refer users to sites or service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. []

UC on Steroids: Incredible PBX for Issabel Joins the Cloud



We’re pleased to introduce the 2019 edition of Incredible PBX® for Issabel featuring new VPS cloud provider support and one-minute setup for Skyetel SIP trunking. One of the limitations of Issabel 4 has been the required use of the ISO installer to deploy Asterisk® 13. This 2019 release addresses that limitation and lets you do a fully scripted install using one of our four recommended $7 to $15 a year VPS cloud providers.

This new release includes our next generation Incredible PBX 13 platform with a preconfigured Travelin’ Man 3 firewall, additional text-to-speech engines (FLITE, GoogleTTS, PicoTTS, and IBM TTS), voice recognition with IBM’s state-of-the-art STT engine, turnkey trunks and extensions, SMS messaging, telephone reminders, turnkey fax support, an AsteriDex phone book with both voice and speed dialing, Wolfram Alpha, sample ODBC apps, and a boatload of dialplan code and AGI scripts to help anyone wanting to learn how to develop custom applications with Asterisk. This is one fantastic UC platform!



Installing Issabel on a Cloud-Based VPS Platform

If you wish to install Issabel 4 on a cloud-based OpenVZ server, here’s the drill. Start by creating a CentOS 7/64 platform. Once the platform is ready, log in to your server as root and immediately change your root password. Then execute the remaining commands in the order listed below. Don’t worry if you cannot access the Issabel web GUI when the install finishes. We’ll fix this up during the Incredible PBX install shortly. Now jump down to the Incredible PBX installation steps to continue.

passwd
yum -y install wget nano
wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash
yum -y erase asterisk
yum -y install asterisk13
reboot

Installing Issabel with Asterisk 13.22.0 from ISO

If you’re using your own hardware or a platform that lets you upload an ISO and deploy, begin by downloading the October 2, 2018 Issabel ISO from SourceForge. On the platform of your choice, install Issabel 4 specifying your Keyboard and Installation Destination with Asterisk 13 as your Software Selection. Add the Sangoma WANPIPE component if desired. Set your Root password and have a cup of coffee. After a reboot, you’ll be prompted to set your MySQL/MariaDB root password (must be passw0rd with a zero) and the admin password of your choice to login to the Issabel web GUI. Be sure to use the new October 2018 Issabel ISO for the base Issabel install. It includes support for Asterisk 13.22.0. We will update things from there as part of the new Incredible PBX install below.

Installing Issabel with VirtualBox

For those using VirtualBox, we’ve uploaded a new Issabel 4 .ova image to SourceForge which will save you some time in getting Issabel up and running. Once you’ve downloaded and installed the image in VirtualBox, you can log in as root using the default password: password. Then you can set your admin password for the Issabel GUI by running /root/admin-pw-change.

Installing Incredible PBX 13 for Issabel 4



As with all Incredible PBX builds, running the Incredible PBX installer will erase ALL of your existing Issabel configuration so start with a fresh install of Issabel.

Begin the Incredible PBX install by logging into your Issabel server as root from a desktop PC using SSH or Putty and execute the following commands:

cd /root
wget http://incrediblepbx.com/IncrediblePBX13-Issabel4.sh
chmod +x IncrediblePBX13-Issabel4.sh
./IncrediblePBX13-Issabel4.sh

The Travelin’ Man 3 firewall is installed and configured as part of the install. It whitelists certain IP addresses and blocks everyone else from even seeing your server on the Internet. For this reason, it is critically important that you perform the Incredible PBX install using SSH or Putty from a PC that you will use to manage your Issabel server. Otherwise, you risk locking yourself out of your own server. Whitelisted IP addresses include the Issabel server itself, the public and private IP addresses of your desktop PC, all non-routable, private LAN addresses, and the Nerd Vittles collection of recommended SIP hosting providers. You can add as many additional providers or users to the whitelist using the simple tools provided as part of the install and further documented below.

As part of the install process, you’ll be prompted during both passes to create a password for MySQL/MariaDB and an admin password for the Issabel web GUI. The MySQL password MUST be passw0rd (with a zero), or you will get a permanent mess. The admin password can be anything you like. Passwords can be updated by running /root/admin-pw-change. Many of the Incredible PBX apps depend upon this MySQL password so don’t change it. Your MySQL databases remain secure and can only be accessed on localhost or after a successful root login to your server from a whitelisted IP address.

WhiteListing IP Addresses in Fail2Ban

We also strongly recommend that you whitelist the IP addresses of computers you plan to use to access your new Issabel PBX. The reason is because Fail2Ban jails take precedence over IPtables settings. So even if your IP address has been whitelisted with IPtables using the Travelin’ Man 3 utilities, it’s still possible to lock yourself out of your server by entering the root or admin passwords incorrectly. Here’s how to avoid that. Edit /etc/fail2ban/jail.conf. Scroll down to line #50 which begins with the word "ignoreip." WhiteListed IP addresses are entered here with a space separating each entry. Once you have entered one or more addresses, save the file. Then restart Fail2Ban: service fail2ban restart.


Introducing the (new) Travelin’ Man 3 Firewall

Issabel 4 includes an IPtables firewall component. Do NOT activate it because Incredible PBX includes its own preconfigured IPtables firewall, better known as Travelin’ Man 3. With the Issabel 4 firewall, the administrator is responsible for setting all of the firewall rules. With Travelin’ Man 3, all the heavy lifting is done for you. The design is also markedly different. Issabel 4 opens ports which you define, but it gives worldwide access to those ports by any user. Travelin’ Man 3 employs a WhiteList rather than opening ports for everyone. If you’re on the WhiteList, you get access to the limited collection of ports assigned to that IP address. If you’re not on the WhiteList, you cannot even see the Issabel PBX from the Internet. For those without remote telephones or traveling employees, this provides total protection of your server with virtually no further firewall management.

If you have remote users of your PBX or if you wish to deploy softphones on mobile devices and rely upon WiFi facilities at random locations, Travelin’ Man 3 provides several utilities to assist. If the remote users have static IP addresses, then those IP addresses can be added to the WhiteList by running /root/add-ip. Better yet, a NeoRouter VPN is provided that lets remote users access Issabel using NeoRouter private LAN addresses that already are WhiteListed as part of the installation process. These require little to no configuration with static or dynamic IP addresses even when switching between WiFi networks. For those with dynamic IP addresses and no VPN, FQDNs can be assigned using a service such as dyn.com and a dynamic DNS client can be loaded on the smartphone to keep the current IP address synchronized with the FQDN. On the Incredible PBX side, these FQDNs can be added using /root/add-fqdn, and the IP addresses will be updated automatically every 10 minutes. The final option to provide remote users the 3-digit PortKnocker codes from knock.FAQ and let them automatically whitelist their own IP addresses by running the PortKnocker client from any smartphone or Linux server. When the Issabel server detects a successful knock sequence, the source IP of the knock sequence is whitelisted until the next reload of the firewall. If an administrator prefers to allow permanent additions to the WhiteList that survive a reboot or restart of the firewall, the administrator need only run the following command one time: iptables-knock activate. WhiteListed entries can be removed using the /root/del-acct utility. Further details on the new Travelin’ Man 3 design are available here.

We have modified the security methodology to access the AsteriDex and Reminders pages in the web GUI. We have added another layer of security by requiring Apache htaccess credentials before you can access these pages on your Issabel server. What this means is you will be prompted for Apache admin credentials when you attempt to access these pages. As the last step of the Incredible PBX installation procedure, you will be asked to specify your admin password again. This becomes your Apache admin password, and we recommend keeping it the same as your Issabel password so you don’t get confused. In this way, the username admin and the admin password will be used BOTH for Apache authentication AND Issabel GUI authentication. Should you ever need to change your Issabel admin password, run /root/admin-pw-change. You will need to execute the following command to change the Apache admin password: htpasswd -c /etc/pbx/wwwpasswd admin.

Overview of Issabel 4 Configuration Steps

Almost all PBXs employ a similar design to get calls flowing in and out of your PBX. Extensions are the hooks that let phones on your PBX make a connection to the PBX. Trunks are the hooks that connect your PBX to the outside world so that you can make and receive external calls. Inbound routes tell the PBX how to route incoming calls from the outside world. Outbound routes tell the PBX which trunk providers to use for various types of outgoing calls. And trunk providers are outside businesses that let you terminate calls to telephones all over the world. They also provide phone numbers (DIDs) to you so that the rest of the world has a way to call you.




Incredible PBX for Issabel makes configuring your PBX easy enough for a fifth grader. We’ve provided two extensions (501 and 502) to give you a simple way to connect your first two phones. We’ve also provided over a dozen sample trunk setups to make it easy to set up trunks once you’ve registered with one or more providers of your choice. If you choose to use our Platinum Sponsor, Skyetel, their trunk setup is already activated and whitelisted on the Issabel platform so all you’ll need to do is collect your $50 signup credit, enter the IP address of your PBX as a Skyetel EndPoint, pick a phone number for your PBX, and point that phone number to your PBX endpoint. On the Issabel side, simply create an Inbound Route for your Skyetel calls by specifying the 11-digit phone number to associate with the inbound route. Finally, we’ll revise the Default Outbound Route to send outgoing calls out through Skyetel.

Getting Started with a $50 Skyetel Credit

To take advantage of the Nerd Vittles specials, begin by completing 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 a $50 credit for your account by referencing the Nerd Vittles special offer. Credit is limited to one per person/company/address/location. If you want to take advantage of the 10% discount on your current service, open another ticket and attach a copy of your last month’s bill. See footnote 1 for the fine print.1 If you have high call volume requirements, document these in your Prequalification Form, and we will be in touch. More details here. Effective 10/1/2023, $25/month minimum spend required.

Skyetel Endpoint Group Configuration

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

  • Name: Issabel
  • Priority: 1
  • IP Address: Issabel-Public-IP-Address
  • Port: 5060
  • Protocol: UDP
  • Description: issabel.incrediblepbx.com

Skyetel DID Configuration

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

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

Incredible PBX Inbound Routing with Skyetel

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

Incredible PBX Outbound Routing to Skyetel

If Skyetel will be your primary provider, you can use both 10-digit and 11-digit dialing to process outbound calls through your Skyetel account. From the GUI, choose PBX -> PBX Configuration -> Outbound Routes -> Default. Scroll down to the Trunk Sequence section of the template. Choose these 3 trunks in this order: Skyetel-1, Skyetel-NW, and Skyetel-SE. Next, click Submit Changes and reload the dialplan when prompted.




Setting Up a Softphone with Issabel 4

If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. You can decipher your server’s IP address by running pbxstatus. If you wish to use one of the preconfigured extensions (501 and 502), you’ll find the randomized passwords in /root/passwords.FAQ. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (D-E-M-O) to try things out. With Telephone, you can use over two dozen soft phones simultaneously.

For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Issabel extension. You’ll need the IP address of your server plus your extension number and password associated with either the 501 or 502 extension.

Adding Speech Recognition Support to Incredible PBX

To support many of our applications, Incredible PBX has included Google’s speech recognition service. These applications include AsteriDex Voice Dialing by Name (411) and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for "personal and development use." If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!

If you like Siri, you’ll love Wolfram Alpha. To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Don’t change the yourID portion of the command:

sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747

Now you’re ready to try out the speech recognition apps. Dial 411 and say "American Airlines" to be connected to American.

To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. "What planes are overhead now?" Read the Nerd Vittles tutorial for additional examples and tips.

Implementing IBM TTS and Voice Recognition

While Google voice recognition originally was free, it has been a hit and miss platform for the last couple years. If you’re really serious about text-to-speech (TTS) and voice recognition (STT) quality, then you owe it to yourself to make the switch to the IBM platform. For most deployments, the IBM platform will be nearly free. Our recent tutorial will walk you through the process of getting your IBM credentials and setting up the TTS and STT functions with IBM Watson. Be advised that you will have two sets of credentials, one for TTS applications and another for STT applications. Once you have your credentials, here are the steps to reconfigure Issabel to use the IBM TTS and voice recognition services. Begin by logging into your server as root and switching to the /var/lib/asterisk/agi-bin directory. Then install the IBM components:

cd /var/lib/asterisk/agi-bin
wget http://incrediblepbx.com/ibm-issabel.tar.gz
tar zxvf ibm-issabel.tar.gz
rm -f ibm-issabel.tar.gz
mv custom/* /var/lib/asterisk/sounds/custom

Implementing IBM STT with Incredible PBX’s Voice Dialer. With this application, a user dials 411 and speaks the name of a person or company to call. The app searches for a match in the AsteriDex directory and places the call. To get started, edit getnumber.sh and insert your IBM STT credentials in the API_USERNAME and API_PASSWORD fields. Then save the file. Replace the Call by Name context by running the following script: ./install-ibm411.sh. Place a test call by dialing 411 and saying "American Airlines."

Implementing IBM STT with Incredible PBX’s SMS Dictator. With this application, a user dials 767, enters the 10-digit number for the recipient of an SMS text message, and then speaks the message to be sent. To get started, edit smsgen.sh and insert your IBM STT and Google Voice credentials using your plain-text Google password. Then save the file. Replace the SMS Dictator context by running the following script: ./install-sms767-dialplan.sh. Place a test call to 767, and the app will send your text message to the recipient’s phone number using the gvoice application. If you experience failed calls, try executing the Unlock Captcha procedure using your Google Voice credentials. Then try again.

Implementing IBM STT with Incredible PBX’s Wolfram Alpha. With this Siri-like app, a user dials 4747 and speaks a query to be sent to Wolfram Alpha for processing. The results then are played back to the caller. To begin, edit wolfram.sh and insert your IBM STT credentials as well as your Wolfram Alpha APPID. Then save the file. Replace the Wolfram Alpha dialplan code by running the following script: ./install-wolfram4747-dialplan.sh. Place a test call by dialing 4747. When prompted for your query, say "What planes are flying overhead now?"

Implementing IBM TTS with Incredible PBX’s News and Weather Apps. With these apps, a user dials 951 for the latest News Headlines from Yahoo or 947 to retrieve the latest weather report by ZIP code. To begin, edit ibmtts.php and insert your IBM TTS credentials in the IBM_username and IBM_password fields. Then save the file. Replace the news and weather by zip code contexts by running the following script: ./install-ibmtts-dialplan.sh.

Generating IBM Voice Prompts to Use with Issabel. We’ve included a script that will let you generate IBM voice prompts that are suitable for use with Issabel and Incredible PBX. To begin, edit ibmprompt.php and insert your IBM TTS credentials in the IBM_username and IBM_password fields. Then save the file. Next, we need to add MP3 support to the SOX application before we can create voice prompts reliably with IBM’s Bluemix TTS service. Here’s how:

yum -y remove sox
yum -y install libmad libmad-devel libid3tag libid3tag-devel lame lame-devel flac-devel
cd /usr/src
wget https://sourceforge.net/projects/sox/files/sox/14.4.2/sox-14.4.2.tar.gz
tar zxvf sox-14.4.2.tar.gz
rm -f sox-14.4.2.tar.gz
cd sox*
./configure
make -s
make install
ldconfig
ln -s /usr/local/bin/sox /usr/bin/sox

Generate voice prompts using the following syntax: ./ibmprompt.php "Hello world."

Configuring the Issabel Fax Server

Incredible PBX for Issabel includes turnkey fax support with Issabel. Once you have added a trunk that supports VoIP faxing (HINT: Skyetel trunks work great!), fax configuration with Issabel only takes a minute. Start by logging into the Issabel web interface as admin. First, navigate to PBX:PBX Configuration:Extensions:Fax and obtain your password for extension 329. Next, navigate to Fax:Virtual Fax:New Virtual Fax. Fill in the form as shown below using your actual email address and phone number for receiving faxes as well as your actual extension 329 secret. Then click SAVE. Assuming you typed your secret correctly, you will see a status notification showing virtual fax machine "Running and idle on ttyIAX1."



Assuming you already have set up a Skyetel trunk as outlined above, the next step is to modify the Inbound Route for this trunk to support fax detection. In that way, incoming fax calls will automatically be redirected to extension 329 and the received faxes will be emailed to you in PDF format. Set the email address in Fax:Fax Master. In addition, the faxes can be downloaded and managed from Fax:Virtual Fax:Fax Viewer. Modify your Inbound Route to match the #3 settings shown below. Then save/reload your changes.



To receive the incoming faxes by email, navigate to Fax:Fax Master and enter your email address. Then click SAVE.

The final step is to designate the IP addresses of those authorized to send faxes using Issabel. Navigate to Fax:Fax Clients and specify the public and private IP addresses (one per line) authorized to send faxes. Then click SAVE. Hylafax clients can be used remotely, or you can use the web utility included with Issabel: Fax:Virtual Fax:Send Fax.




The best way to test things out is to send yourself a test fax. FaxZERO lets you send 5 free faxes of up to 3 pages every day. Give it a whirl.

To send a fax out from your server from the Linux CLI using either a text document or PDF file, the syntax looks like the following:

sendfax -n -d 8005551212 smsmsg.txt



Replacing MeetMe Conferencing with ConfBridge

The only serious limitation we’ve found with the Issabel implementation of FreePBX is the continued reliance upon MeetMe for conferencing which requires a timing source unlike the newer ConfBridge module. Particularly on OpenVZ VPS platforms, this causes issues because of the inability to directly access the kernel. Fortunately, Issabel has included the functioning ConfBridge module in their implementation so the workaround is fairly simple. By default, we’ve included a 2663 (C-O-N-F) conference setup in the Issabel GUI configuration so simply remove it. Then add a 2663 Misc Destination with a description of CONF. Finally, while still in the GUI, edit the IVR Demo and change the destination for option 2 to Misc Destination:CONF and save the file. Next, log into the Linux CLI as root and change to the /etc/asterisk directory. Edit confbridge_custom.conf and insert the following code. Then save the file.

[general]
;This section reserved for future use

[default_user]
type = user
quiet = no
announce_user_count = yes
announce_user_count_all = yes
wait_marked = no
end_marked = no
dsp_drop_silence = yes
announce_join_leave = yes
admin = no
marked = no
startmuted = no
music_on_hold_when_empty = yes

[admin]
type = user
quiet = no
announce_user_count = yes
announce_user_count_all = yes
wait_marked = no
end_marked = no
dsp_drop_silence = yes
announce_join_leave = yes
admin = yes
marked = no
startmuted = no
music_on_hold_when_empty = yes

[default_bridge]
type = bridge
record_conference = no
sound_only_person =    conf-onlyperson
sound_has_joined =     conf-hasjoin
sound_has_left =       conf-hasleft
sound_kicked =         conf-kicked
sound_muted =          conf-muted
sound_unmuted =        conf-unmuted
sound_there_are =      conf-thereare
sound_other_in_party = conf-otherinparty
sound_place_into_conference = conf-placeintoconf
sound_wait_for_leader =       conf-waitforleader
sound_get_pin =        conf-getpin
sound_invalid_pin =    conf-invalidpin
sound_locked =         conf-locked
sound_unlocked_now =   conf-unlockednow
sound_lockednow =      conf-lockednow
sound_error_menu =     conf-errormenu

[admin_menu]
type = menu
* = playback_and_continue(conf-adminmenu)
*1 = toggle_mute
*2 = admin_toggle_conference_lock
*3 = admin_kick_last
*4 = decrease_listening_volume
*5 = reset_listening_volume
*6 = increase_listening_volume
*7 = decrease_talking_volume
*8 = reset_talking_volume
*9 = increase_talking_volume
*# = leave_conference
*0 = admin_toggle_mute_participants

[user_menu]
type = menu
* = playback_and_continue(conf-usermenu)
*1 = toggle_mute
*4 = decrease_listening_volume
*5 = reset_listening_volume
*6 = increase_listening_volume
*7 = decrease_talking_volume
*8 = no_op
*9 = increase_talking_volume
*# = leave_conference

Now edit extensions_custom.conf and insert the following code below the [from-internal-custom] label replacing the 1234 and 4321 PINs in lines 6 and 7 with user and admin PINs of your choice (up to 8 numbers each). Then restart Asterisk: amportal restart.

;# // BEGIN Conf1
exten => 2663,1,Answer
exten => 2663,2,Wait(1)
exten => 2663,3,Playback(conf-getpin)
exten => 2663,4,Read(MYPIN,beep,8)
exten => 2663,5,GotoIf($["${MYPIN}" = "1234"]?userpin)
exten => 2663,6,GotoIf($["${MYPIN}" = "4321"]?adminpin)
exten => 2663,7,Playback(goodbye)
exten => 2663,8,Hangup
exten => 2663,n(adminpin),Set(CONFBRIDGE(user,template)=admin)
exten => 2663,n,ConfBridge(1)
exten => 2663,n,Hangup
exten => 2663,n(userpin),Set(CONFBRIDGE(user,template)=default_user)
exten => 2663,n,ConfBridge(1)
exten => 2663,n,Hangup
;# // END Conf1

Backup and Restore with Issabel

Issabel ships with the most full-featured Backup and Restore options of any of the Asterisk distributions. Ask us how we know. Yes, we managed to wipe out the entire Dashboard menu system on one of our early builds. Restoring from an image took only a couple minutes. To get started, navigate to System -> Backup/Restore. You can create backups locally and then drag and drop them onto a remote FTP server if desired. There is enormous flexibility in choosing what to backup or restore. And there’s even an option to automatically generate periodic backups. You’ll find your backups in /var/www/backup should you ever need to copy them to a new server. Now would be a good time to create your first backup. 🙂

Sampling Other Incredible PBX Applications

As installed, Incredible PBX includes dozens of additional applications for Asterisk. Here’s how to sample some of them using a softphone connected to your Issabel PBX. A good place to start is Allison’s Demo IVR (dial D-E-M-O) using any phone connected to your PBX:

Nerd Vittles Demo IVR Options
1 – 411 -Call by Name (say "American Airlines")
2 – 2663 – MeetMe/ConfBridge Conference
3 – 4747 – Wolfram Alpha
4 – 53669 – Lenny (The Telemarketer’s Worst Nightmare)
5 – 951 – Today’s News Headlines
6 – 947 – Weather Forecast (enter a 5-digit ZIP code)
7 – 86329 – Today in History
8 – 501 – Speak to a Real Person

For ODBC demos, dial 222 and enter 12345 for the employee number for a sample database application. Or dial 223 for a sample ODBC dialer using AsteriDex. Enter 263 (first three letters of American Airlines) to place the call. Sample dialplan code is stored in /etc/asterisk/odbc.conf. Dial L-E-N-N-Y (53669) to call or forward telemarketer calls to Lenny. Dial T-I-M-E (8463) for Time of Day. Dial *88HHMM to set an Alarm for HH:MM where HH is the hour of the day in military time. Dial C-O-N-F (2663) for MeetMe conference. Conference credentials are in /root/passwords.FAQ. Voice Dialer (411) works with any database entry in AsteriDex. Access AsteriDex with a browser at https://Issabel-IP-Address/asteridex4. Telephone Reminders can be scheduled by phone (123) or via the web: https://Issabel-IP-Address/reminders. Sample code for the FLITE, GoogleTTS, and PicoTTS engines is in 951 (Yahoo News) context of /etc/asterisk/extensions_custom.conf. All of your FreePBX "old favorites" including blacklists, call transfers and forwarding, dictation, recordings and more are still available as well: PBX:PBX Config:Feature Codes.

Continue Reading: Configuring Extensions, Trunks & Routes.

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

Published: Friday, October 5, 2018  Updated: Friday, February 1, 2019


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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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

Introducing Skyetel: A VoIP Provider for All Seasons

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

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

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

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

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

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

Skyetel Pricing Overview

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

Signing Up for Skyetel Service

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

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

Skyetel Endpoint Group Configuration

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

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

Skyetel DID Configuration

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

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

Incredible PBX Firewall Setup for Skyetel

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

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

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

Incredible PBX Trunk Setups for Skyetel

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

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

Incredible PBX Inbound Routing for Skyetel

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

Incredible PBX Outbound Routing to Skyetel

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

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

Audio Issues with Skyetel

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

externip=xxx.xxx.xxx.xxx

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

Receiving SMS Messages Through Skyetel

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



Sending SMS Messages Through Skyetel

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

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

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

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

SMS and MMS Messaging with Postcards

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

NEW: Skyetel Support for Dynamic IP Addresses

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

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

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

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

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

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

apt-get -y install coreutils curl git jq

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

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

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

Introducing Skyetel’s New Fax Platform

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


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

Implementing the New Spam Call Filter

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

Recording and Transcribing Skyetel Calls

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

Skyetel Monitoring of Endpoint Health

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

Skyetel Expansion for Canadian Users


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

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

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


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



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



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

SIP Happens! And Kamailio Makes It Easy, Part I




If ever there was a Swiss Army Knife for SIP, Kamailio (a.k.a. OpenSER) is the hands-down winner. The flexibility of this open source SIP server is legendary. Whether it’s secure communications, insulation from brute force attacks, load balancing, failover, WebRTC, or the return of shared line appearances on your office phone system, Kamailio can handle it while processing thousands of call setups per second on minimal hardware platforms.

Our plan for today is to walk you through setting up a Debian-based Kamailio server on an inexpensive cloud platform that is suitable for making thousands of free SIP phone calls worldwide. Down the road, we’ll walk you through using Kamailio as a frontend for one or more Asterisk® servers to insulate your communications workhorses without sacrificing network security. If, like us, you managed an office which migrated from key telephones to a platform like Asterisk, then you will most certainly appreciate the ability to once again let your managers and secretaries share phone lines without the aggravation of call parking and pickup. Other than removing a free office coffee machine, I can’t think of any single event that ever prompted a staff and management revolt quite like the one we experienced with the removal of key telephones. Little wonder that it’s part of all Cisco and Avaya phone systems as well as cloud offerings from Vonage, 8X8, Jive, and probably others.

Before we begin our adventure, let me caution everyone that this is an experimental platform with a tutorial prepared by a Kamailio novice. While we’ve done our homework, digging out information on Kamailio is a challenge because many of the experts depend upon Kamailio consulting for their livelihood. It’s quite similar to the early Asterisk years. We also don’t vouch for the longevity of any of these VPS providers. Reread our article for details.

SIP URI (Free) Calling Opportunities

We mentioned free SIP phones in our introduction of Kamailio. But let that sink in for a moment. As we have stressed for many years, SIP calls to anyone with a public SIP URI (somebody@somewhere.com) are entirely free to anyone worldwide. And you can talk as long as you like. All that’s required is an Internet connection, a SIP phone or softphone, and a SIP URI. As part of the Kamailio implementation, we’ll show you how easy it is to create SIP URIs for all your friends and business acquaintances securely… in seconds. First, let’s take a moment to consider what SIP URI (free) calling opportunities are available. There literally are millions of SIP URI resources that await. But, unless you want to be one of the "don’t call us, we’ll call you" folks, you also will need one or more SIP URIs for yourself. YOU DON’T NEED A KAMAILIO SERVER TO OBTAIN A SIP URI. Here are just a few of the possibilities. Using SIP Broker, you can call anyone on more than 2,000 VoIP networks around the globe. Using a softphone and a free or almost free registration with VoIP.ms, CallCentric, or LocalPhone, you not only get a SIP URI, but you also can request an iNUM number which also doubles as a SIP URI by coupling it with @81.201.82.50. 3CX and pbxes.org also offer SIP URIs to complement their free offerings. All of these companies will let you connect a softphone or SIP endpoint directly to their service without the need for an Asterisk PBX in the middle.1 You can even refer your favorite spam callers to Lenny via SIP URI: 2233435945@sip2sip.info.

Deploying a Cloud-Based Debian Server

We hope you took advantage of one of the special VPS offerings we highlighted to start the New Year. Some are still available with annual pricing that’s less than the cost of most lunches these days. We recommend a cloud platform because (1) it’s cheap, (2) it’s easy to set up a Debian platform, and (3) it provides a static IP4 address for your server. When you sign up or if you wish to reconfigure an existing VPS that you may have gathering dust, just choose the Debian 8/64 operating system and assign an FQDN to your server. Once you get your credentials, log into the server as root with the password that was provided. Immediately change your root password and issue the following commands to bring Debian up to date. We also strongly recommend changing the SSH port to deter would-be attackers. A TCP port in the 1000-2000 range works wonders. Simply edit /etc/ssh/sshd_config and change the Port 22 entry before rebooting. HINT: Birth years make the SSH port easy to remember.

passwd
apt-get update
apt-key update
apt-get dist-upgrade
apt-get -y install gcc flex bison libmysqlclient-dev make libssl-dev nano
apt-get -y install libcurl4-openssl-dev libxml2-dev libpcre3-dev ntp ntpdate
reboot

Once the reboot is complete, log back into your server’s new SSH port using this syntax where 1234 is the port number you chose.

ssh -p 1234 root@server-ip-address

Now we’re ready to install the necessary packages to support Kamailio:

apt -y install mysql-server
mysql_secure_installation
apt -y install kamailio kamailio-mysql-modules
apt -y install kamailio-dbg
apt -y install kamailio-extra-modules
apt -y install kamailio-outbound-modules
apt -y install kamailio-presence-modules
apt -y install kamailio-tls-modules
apt -y install kamailio-utils-modules
apt -y install kamailio-websocket-modules

Configuring Kamailio’s kamctlrc File

For today, we’ll be configuring Kamailio to allow user logins from SIP endpoints including SIP phones and softphones. Down the road, we’ll change things up to let Kamailio serve as the front-end to one or more Asterisk PBXs. But let’s learn to walk before we start running. We’ll be editing three configuration files and then adding a SIP account to support logging in from a SIP phone. Let’s begin with kamctlrc.

(1) Edit kamctlrc: nano -w /etc/kamailio/kamctlrc

(2) Start by uncommenting SIP_DOMAIN and insert the FQDN you associated with your VPS.

(3) Uncomment DBENGINE line and make certain it points to MYSQL.

(4) Uncomment the following line: DBRWUSER="kamailio".

(5) Uncomment the DBRWPW line and insert your own password between the quotes.

(6) Uncomment the following line: DBROUSER="kamailioro".

(7) Uncomment the DBROPW line and insert a different password between the quotes.

(8) Uncomment the DBACCESSHOST line and insert the IP address of your server.

(9) Drop down near the bottom of the file and uncomment the PID_FILE line.

(10) Save the file.

Configuring Kamailio’s kamailio.cfg Startup File

(1) Edit kamailio.cfg: nano -w /etc/kamailio/kamailio.cfg

(2) Make the top of the startup file look like the following:

#!KAMAILIO
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_USRLOCDB
#!define WITH_ANTIFLOOD
#!define WITH_PRESENCE
# change next line to comment to disable logging
#!define WITH_ACCDB
#
# Kamailio (OpenSER) SIP Server v4.2 - default configuration script

(3) Find the line: #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"

(4) Change the kamailiorw entry to the password you entered in step #5 above.

(5) Tighten up security a bit by searching for the line containing friendly-scanner.

(6) Immediately above that line, cut-and-paste this addition from Fred Posner at AstriCon:

### Posner additions
        if ($ua =~ "(friendly-scanner|sipvicious|sipcli)") {
                xlog("L_INFO","script kiddies from IP:$si:$sp - $ua \n");
$sht(ipban=>$si) = 1;
                sl_send_reply("200", "OK");
                exit;
        }
        if($au =~ "(\=)|(\-\-)|(')|(\#)|(\%27)|(\%24)" and $au != $null) {
                xlog("L_INFO","[R-REQINIT:$ci] sql injection from IP:$si:$sp - $au \n");
$sht(ipban=>$si) = 1;
                exit;
        }
###

(7) Save the file.

(8) Generate the MySQL database and tables to support Kamailio: kamdbctl create

(9) At every prompt, type Y to add the feature.

(10) Open MySQL as root using the actual MySQL password you assigned when adding the MySQL package:

mysql -u root -ppassw0rd kamailio

(11) At the MySQL prompt, cut-and-paste the following commands:

ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';
ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';
ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';
quit

Configuring Kamailio Defaults in /etc/default/kamailio

(1) Edit Kamailio defaults: nano -w /etc/default/kamailio

(2) Make the startup defaults look like the following:

#
# Kamailio startup options
#

# Set to yes to enable kamailio, once configured properly.
RUN_KAMAILIO=yes

# User to run as
USER=kamailio

# Group to run as
GROUP=kamailio

# Amount of shared and private memory to allocate
# for the running Kamailio server (in Mb)
SHM_MEMORY=128
PKG_MEMORY=4

# Config file
CFGFILE=/etc/kamailio/kamailio.cfg

(3) Save the file.

Managing Kamailio Startups & Shutdowns

With all the pieces in place, here’s how to start, restart, stop, and check status of Kamailio:

systemctl start kamailio
systemctl restart kamailio
systemctl stop kamailio
systemctl status kamailio

Adding Users/Accounts to Kamailio

Now we’re ready to add accounts to Kamailio. These can be numeric, alphanumeric, or purely alpha entries. They become the user’s respective SIP URIs when coupled with @FQDN where FQDN is the fully-qualified domain name assigned to your server:

kamctl add username userpw

As you probably have guessed, kamctl is the main management tool for Kamailio. Issuing the command by itself will list all of the possible options that are available.

Monitoring Kamailio Access

There are a number of ways to monitor access (both legitimate and otherwise) to your Kamailio server. Here are a few of our favorites:

systemctl status kamailio
cat /var/log/syslog | grep kamailio
mysql -u root -ppassw0rd kamailio -e "select * from acc"
mysql -u root -ppassw0rd kamailio -e "select * from missed_calls"

Connecting a SIP Phone to Kamailio

You can connect virtually any kind of SIP telephone or endpoint to Kamailio. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum. For today we’ll get you started with one of our favorite (free) softphones, YateClient. It’s available for almost all desktop platforms.

Download YateClient from here. Run YateClient once you’ve installed it and enter the credentials for the account you created above. You’ll need the IP address of your server plus your account’s password. Fill in the Yate Client template using the IP address or FQDN of your Server as well as your Username and whatever Password you assigned to the account when you created it. Click OK to save your entries.

Once the Yate softphone shows that it has registered with Kamailio, try a test call to Lenny by dialing sip:2233435945@sip2sip.info.

Next week, we’ll tackle security. If you run systemctl status kamailio for a few days, you’ll understand why. We’ll also get your Kamailio server interconnected with Asterisk so that inbound calls to your new SIP URI pass through to Asterisk transparently. Enjoy!

Originally published: Monday, January 14, 2019



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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



  1. Some of our links refer users to sites or service providers when we find their prices are competitive for the recommended products. Nerd Vittles receives a small referral fee from these providers to help cover the costs of our blog. We never recommend particular products solely to generate commissions. []

A Sobering Look at Asterisk and the 2019 VoIP Landscape




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

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

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

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

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

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

Originally published: Monday, December 17, 2018



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


 

Special Thanks to Our Generous Sponsors


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

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

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

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

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