Category Archives: Smartphones

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

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.

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!

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

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

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.

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

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.

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.

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.

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.

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.

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:

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"

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.

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.

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.

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."

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.

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.

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.

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.

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.

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.

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