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Incredible PBX 13-13 Application User’s Guide

For those just beginning the Incredible PBX® 13-13 adventure, start here. Once your system is up and running, you’ll be ready to kick the tires. And today we’ll cover the applications for Asterisk® that are included in the latest and greatest Incredible PBX Whole Enchilada. If you have questions, post them on the PIAF Forum for some quick and friendly assistance.

Here’s a Table of Contents to the Incredible PBX 13-13 Applications with hotlinks. Enjoy!

  1. Checking System Status
  2. Enabling Speech Recognition for Asterisk
  3. Wolfram Alpha for Siri-like queries by phone*
  4. Automatic Update Utility
  5. Resetting Incredible PBX Passwords
  6. Apache Authentication for Apps
  7. IPtables Firewall WhiteList
  8. PortKnocker Remote Access
  9. Travelin’ Man 4 Remote Access by Phone
  10. Conference Bridge
  11. CallerID Name (CNAM) Lookups
  12. Faxing with Incredible PBX
  13. Voicemail 101 with Incredible PBX
  14. Email Delivery of MP3 Voicemails
  15. Reconfiguring SendMail for SmartHosts
  16. SMS Blasting with Google Voice
  17. SMS Voice Messaging with Google Voice*
  18. SMS Messaging with VoIP.ms
  19. SIP URI Calling with Speed Dials
  20. IVR Demo of Incredible PBX Applications*
  21. Backup and Restore Options
  22. AsteriDex – The Poor Man’s Rolodex®
  23. Voice Dialing with AsteriDex*
  24. Speed Dialing with AsteriDex
  25. Scheduling Reminders by Phone or Web
  26. DISA Access with Incredible PBX
  27. Yahoo! News Headlines
  28. Weather Forecasts with Incredible PBX*
  29. ODBC Application Support
  30. Today in History
  31. Time of Day

* Requires Voice Recognition implementation. See #2 above.

1. Checking Current Status of Incredible PBX

There are several ways to check the status of your server. First, log into your server as root and type: status or pbxstatus. You can even add the default phone number for your server by inserting it in /etc/pbx/.phone.

The second option is to use a browser to access your server. Choose the Incredible PBX Admin option after pointing a browser to the IP address of your server:

Once you log in with your admin password, the Dashboard of your server will display the status of trunks, users, and active calls on your server. In addition, you can review the latest news and security alerts from the RSS Feeds of Nerd Vittles, Incredible PBX, FreePBX®, and Asterisk. For additional status information, choose Reports:Asterisk Info.

2. Adding Speech Recognition to Asterisk

Google changed the licensing of their speech recognition engine last year and now restricts use to "personal and development use." Assuming you qualify, the very first order of business is to enable speech recognition for your new PBX. We no longer recommend Google Speech Recognition because of the licensing issues and Google’s propensity to break things regularly. Instead, we recommend IBM’s Speech Recognition and TTS engines. For most users, there will be no cost. And the services are second to none. For a complete installation and setup tutorial, see our original article on Incredible PBX 13-13. Once speech recognition is enabled, the Incredible PBX feature set grows exponentially. You’ll have access to the Voice Dialer for AsteriDex as well as SMS Voice Messaging and Wolfram Alpha for a Siri-like encyclopedia.

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

Here’s how to activate Google speech recognition on Incredible PBX if you choose not to use the IBM solution. Don’t skip any steps!

1. Using an existing Google/Gmail account, you first must join the Chrome-Dev Group.

2. Using the same account, create a new Speech Recognition Project.

3. Click on your newly created project and choose APIs & auth.

4. Turn ON the Speech API by clicking on its Status button in the far right margin. HINT: If you forgot to complete Step #1, the Speech API option will be missing!

5. Click on Credentials in APIs & auth and choose Create New Key -> Server key. Leave the IP address restriction blank!

6. Write down your new API key or copy it to the clipboard.

7. Log into your server as root and issue the following command:

nano -w /var/lib/asterisk/agi-bin/speech-recog.agi

8. When the nano editor opens, go to line 70 of speech-recog.agi: my $key = "". Insert your API key from Step #6 above between the quotation marks and save the file: Ctrl-X, Y, then Enter.

3. Using Wolfram Alpha with Incredible PBX

Ever wished your Asterisk server could harness the power of a 10,000 CPU Supercomputer to answer virtually any question you can dream up about the world we live in? Well, so long as it’s for non-commercial use, today’s your lucky day. Apple demonstrated with Siri™ just how amazing this technology can be by coupling Wolfram Alpha® to a speech-to-text engine on the iPhone. Now you can do much the same thing using voice recognition on the Incredible PBX for Asterisk-GUI.

Before using Wolfram Alpha from any phone connected to your PBX, you first must configure it by obtaining and adding a Wolfram Alpha application ID to Incredible PBX. Here are the simple steps:

1. Obtain your free Wolfram Alpha APP-ID here.

2. Log into your server as root and issue the following command:

nano -w /var/lib/asterisk/agi-bin/wolfram.sh

3. When the nano editor opens, insert your IBM STT and Wolfram APP-ID credentials in the spaces provided. Then save the file.

To use Wolfram Alpha, dial 4747 (that’s S-I-R-I backwards) from any extension.

Here are some sample queries to get you started:

Weather in Charleston South Carolina
Weather forecast for Washington D.C.
Next solar eclipse
Otis Redding
Define politician
Who won the 1969 Superbowl? (Broadway Joe)
What planes are flying overhead now?
Ham and cheese sandwich (nutritional information)
Holidays 2015 (summary of all holidays for 2015 with dates and DOW)
Medical University of South Carolina (history of MUSC)
Star Trek (show history, air dates, number of episodes, and more)
Apollo 11 (everything you ever wanted to know)
Cheapest Toaster (brand and price)
Battle of Gettysburg (sad day 🙂 )
Daylight Savings Time 2015 (date ranges and how to set your clocks)
Tablets by Samsung (pricing, models, and specs)
Doughnut (you don’t wanna know)
Snickers bar (ditto)
Weather (local weather at your server’s location)

4. Automatic Update Utility for Incredible PBX

A key security component of Incredible PBX is its Automatic Update Utility. Each time you log into your server as root, the Automatic Update Utility is run. It installs the latest fixes and security patches for your server. Don’t disable it! In fact, don’t delete anything from the /root folder. You’ll need all of it sooner or later.

We recommend you log into your server as root at least once a week to keep your server current. Ditto for the web interface to Incredible PBX. Insofar as security is concerned, we make a best effort to keep the components of Incredible PBX up to date. The Linux operating system was installed by you before the Incredible PBX install began. That’s a nice way of saying Linux security is primarily your responsibility. When an egregious Linux vulnerability comes along that we know about, we will try to notify you of the issue on the PIAF Forum and on the RSS Feed that is part of the Incredible PBX GUI. Check the RSS Feeds at least once a week as well. As a condition of use of the free Incredible PBX product, you accepted ultimate responsibility for the security and reliability of your server. None of this discussion changes any of that.

5. Resetting Incredible PBX Passwords

Yes. It happens to all of us. We forget our passwords. Incredible PBX includes a convenient utility that lets you reset many of the passwords associated with Incredible PBX. Just log into your server as root and issue the command: /root/update-passwords

To reset Incredible PBX GUI admin password, issue command: /root/admin-pw-change

To reset the AvantFax admin password which is accessible within the Incredible PBX GUI, issue the following command: /root/avantfax-pw-change

6. Apache Authentication with Incredible PBX

With the exception of the Admin GUI and WebMin, all web-based applications included in Incredible PBX require successful Apache authentication to gain access. When you installed Incredible PBX, you should have created an admin account for Apache. If not, issue the following command using a secure password after logging in as root:

htpasswd -cb /etc/pbx/wwwpasswd admin newpassword

With the exception of AsteriDex and Reminders, you gain access to other Incredible PBX applications with the admin Apache account. For the remaining apps, you may wish to (but don’t have to) assign different account names and passwords to various departments in your organization. To set up these accounts, use the syntax above substituting the name of the department for "admin" and the department password for "newpassword."

7. Managing the IPtables Linux Firewall

As installed, Incredible PBX includes a preconfigured, locked-down Linux firewall that restricts incoming IPv6 traffic to localhost and, via a Travelin’ Man 3 WhiteList application, limits incoming IPv4 traffic to your server’s public and private IP addresses, your desktop computer’s IP address (that was used for the install), private LAN and NeoRouter VPN traffic, and a collection of our favorite VoIP providers. You can WhiteList additional IP addresses for additional providers or for SIP and IAX phones located outside your firewall. The following firewall management scripts are accessible from the /root directory:

  • ./add-ip — WhiteList an additional IP address or IP address range (CIDR)
  • ./add-fqdn — WhiteList a site using a fully-qualified domain name (FQDN)
  • ./del-acct — Remove previously designated entry from the WhiteList
  • ./ipchecker — Check whether specified FQDNs have changed & update IPtables
  • iptables-restart — Used exclusively to restart IPtables and test for failed FQDNs
  • iptables -nL — Check the current status of your IPtables firewall

IPtables can be manually configured (if you know what you’re doing) by editing iptables and ip6tables in /etc/sysconfig. Additional IPtables rules are included and managed in /usr/local/sbin/iptables-custom. NEVER use traditional iptables commands such as iptables save to update your IPtables configuration, or you will permanently delete all of your FQDN entries! Instead, use the provided utilities to whitelist additional sites and then restart IPtables using iptables-restart. This protects the FQDN entries in your setup while also checking for invalid FQDN entries and removing them temporarily so that IPtables will successfully restart. If you use service iptables restart to restart IPtables and there happens to be an FQDN entry for a host that is either down or has disappeared, IPtables will fail to restart and your server will be left with NO firewall protection! Using the traditional IPtables mechanisms also will disable Fail2Ban. Incredible PBX periodically checks for changed FQDN entries using the ipchecker script configured in /etc/crontab.

If you elect to integrate Facebook into your Incredible PBX setup, you will need to manually uncomment the last 3 lines in /usr/local/sbin/iptables-custom in order to whitelist the Facebook servers. Then restart the firewall: iptables-restart

WARNING: By default, Incredible PBX whitelists all of the non-routable LAN subnets including 10.0.0.0/8, 172.16.0.0/12, and 192.168.0.0/16. If you elect to install Incredible PBX in the Amazon Cloud, be advised that Amazon treats the 172.16.0.0/12 subnet as routable IP addresses. This means that anyone in the Amazon Cloud (including the bad guys) will have direct access to your server. While they still need a password or vulnerability to gain access, it nevertheless exposes your server to needless hacking attempts. We strongly recommend that you comment out the 172.16.0.0/12 entry in /usr/local/sbin/iptables-custom if you intend to deploy your server in the Amazon Cloud. Then restart the firewall: iptables-restart

8. PortKnocker Remote Access

IPtables is a powerful firewall that keeps the bad guys out. It also will keep legitimate users (including you) from gaining remote access to your server unless you had the forethought to WhiteList your remote IP address before you left on that family vacation. Unfortunately, you don’t always know your IP address in advance. And dynamic IP addresses assigned with hotel WiFi frequently change. To address this problem, Incredible PBX includes a preconfigured PortKnocker utility. This lets you send three secret "knocks" on random TCP ports to your server to tell it to let you in either temporarily (until IPtables is restarted) or permanently.

To reconfigure PortKnocker to permanently whitelist IP addresses from which you issue a successful knock, login as root and issue the command: iptables-knock activate

For PortKnocker to work, you obviously need to know the secret knocks. You’ll find them in /root/knock.FAQ. Record them in your wallet or inside your suitcase for that rainy day! There are PortKnocker apps for almost all smartphones as well as for Windows, Mac, and Linux computers. Install your favorite AND test access before you leave town. You can change the ports by editing /etc/knockd.conf. Then restart PortKnocker: service knockd restart

Finally, be aware that PortKnocker does not need any special access to your server to work; however, if your server is behind a hardware-based firewall, then you must map the three PortKnocker TCP ports to the private IP address of your server, or the knocks obviously will never get delivered to your server.

If you installed Incredible PBX on a cloud platform, then your server may use a network port other than eth0. Typically, it’s venet0:0 on OpenVZ servers. You can decipher the name of your network port for your public IP address by issuing the command: ifconfig. In this case, the config file needs to be modified and then PortKnocker needs to be restarted. Edit /etc/sysconfig/knockd and insert the following: OPTIONS="-i venet0:0". Restart PortKnocker with the command: service knockd restart

Review our PortKnocker tutorial for additional configuration tips.

9. Travelin’ Man 4 Remote Access (dial TM4)

In addition to PortKnocker, Incredible PBX also includes a telephone-based solution to temporarily gain remote access to your server. This does require a bit of preplanning since you must create account credentials for the person to whom you wish to give remote access via a phone call. The complete tutorial for Travelin’ Man 4 is available on the PIAF Forum. All of the pieces already are in place on your server so skip down to the Configuration & Operation sections for details on implementation. The tutorial also covers the Administrator Utilities in /root/tm4 which let you set up remote user accounts.

10. Using the Conference Bridge (dial CONF)

A new turnkey Asterisk 13 Conference Bridge has been added to Incredible PBX. A conference bridge allows a group of people to participate in a joint phone call. Typically, participants dial into a virtual meeting room from their own phone. This virtual meeting room supports dozens or even hundreds of participants depending upon server capacity.

You do not need a timing source for conferencing with Incredible PBX! Old-style Asterisk Conference Rooms which required a timing source are disabled.

To access the Conference Bridge, dial C-O-N-F (2663) from any phone connected to your server. Remote users can be added to a conference by providing a DID that points to an IVR which includes Conference Bridge access. Once connected to the conference bridge, a caller is prompted for the Conference Bridge PIN and his or her name. You can decipher or modify the user and admin passwords to access the Conference Bridge in the Incredible PBX GUI: Applications:Conferences. Then edit 2663 and review the User and Admin PINs.

11. CallerID Name (CNAM) Lookups

By default, Incredible PBX is configured to automatically provide OpenCNAM CallerID name lookups for the first ten calls received each hour. These lookups are only from cached entries in the OpenCNAM database; however, you can enable the commercial lookup service if desired. The cost is four tenths of a cent per successful query.

To enable the OpenCNAM Professional Tier, set up an account at OpenCNAM.com. Once you’ve obtained your credentials, edit the OpenCNAM entry in Admin:CID Superfecta:Default. You may also wish to enable AsteriDex lookups and move the scheme to the top of your list of lookup schemes.

To activate CallerID Superfecta for incoming calls, edit each of your Inbound Routes and Enable Superfecta Lookup with the Default Scheme in the Other tab.

12. Faxing with Incredible PBX

If you can press the ENTER key 25 times, you are fully capable of installing Incredible Fax on your new server. Log into your server as root and run /root/incrediblefax13.sh. Provide an email address for delivery of incoming faxes and press ENTER each time you are prompted to make a selection. Once you reboot your server, you’re all set. As part of the install, you provided an email address for delivery of incoming faxes. That’s all the setup that is required to have incoming faxes sent to most of your DIDs delivered via SendMail in PDF format. The best way to figure out whether a particular provider supports fax technology on their DIDs is to send a test fax to yourself. FaxZERO lets you send 5 free faxes of up to 3 pages every day. Give it a whirl.

You also can send faxes using standard document types with the AvantFax web application. Log into AvantFax from the main Incredible PBX GUI by clicking on the AvantFax icon. The default credentials are admin:password. Choose the Send a Fax option from the main menu, fill in the blanks, and attach your document. AvantFax uses the default dialplan so use the prefix desired to send the fax using your preferred provider. HINT: Google Voice does an excellent job with both incoming and outgoing faxes, and the calls are free in the U.S. and Canada.

With the latest release of Incredible PBX 13-13, fax recognition is supported on incoming calls. Edit each of your Inbound Routes and enable Detect Faxes with Detection Type=SIP, Fax Ring=Yes, Fax Detection Time=4, and Fax Destination=Custom Destination:Fax (HylaFax) in the Fax tab.

Copies of all incoming faxes also are available for retrieval within AvantFax.

13. Voicemail 101 for Incredible PBX

Voicemail functionality is enabled on an extension-by-extension basis as part of the extension setup under the Voicemail tab. Once enabled, you can set up your mailbox and retrieve your messages by dialing *97 from the mailbox extension, or dial *98 to retrieve messages from any extension. Shortcut dialing is also supported, e.g. *98707 would retrieve messages for extension 707. You can leave a message for or forward calls to any extension’s mailbox without actually calling the extension. Just prepend * to any extension number before dialing, e.g. *701. A number of the system settings for voicemail can be tweaked under the Voicemail tab as well. For example, you can automatically delete voicemails once they have been delivered by email. Voicemail Blasting to multiple mailboxes is also supported. Just choose this option under the Applications tab and follow your nose.

14. Email Delivery of MP3 Voicemails

Speaking of email delivery, your voicemails also can be delivered to any email address of your choosing. For every extension under the Voicemail tab for the Extension, simply add an Email Address and enable the Email Attachment. With Incredible PBX, the voicemail message will be attached to the email in MP3 format so it’s suitable for playback with most email clients on desktop PCs, Macs, and smartphones. Be advised that some Internet service providers (such as Comcast) block downstream SMTP servers. You can check whether your outbound email is flowing by accessing WebMin (below) and choosing Servers -> SendMail Mail Server -> Mail Queue. Or you can issue the following command using your destination email address:

echo "test" | mail -s testmessage your-name@your-email-provider.com

If you find outbound mail is accumulating after checking in WebMin, then you’ll need to add your ISP’s SMTP server address as a SmartHost for SendMail as documented in the next section.

15. Reconfiguring SendMail for a SmartHost

Many residential Internet service providers block downstream SMTP servers such as the SendMail server running with Incredible PBX. If you’re sending emails but they never arrive and you’ve checked your SPAM folder, then chances are your ISP is the culprit. The simple solution is to add your ISP’s SMTP server as a SmartHost for SendMail. This means outbound emails will be forwarded to your ISP for actual email transmission over the Internet. Here’s how. Edit /etc/mail/sendmail.cf and search for DS. Immediately after DS, add the FQDN of your ISP’s SMTP server, e.g. DSsmtp.comcrap.net (no spaces!). Save the file and then restart SendMail: service sendmail restart. Your email and voicemail messages with attachments should begin flowing without further delay.

16. SMS Blasting with Google Voice

Out of the box, Incredible PBX supports SMS Message Blasting if you have a functioning Google Voice account set up. Before first use, you must add your credentials, address list, and text message to the SMS Blaster scripts in the /root folder.

In smsblast, insert your credentials:

GVACCT="yourname@gmail.com"
GVPASS="yourpassword"
MSGSUBJECT="Little League Alert"

In smslist.txt, insert one or more recipients for your message. These can be a combination of SMS addresses and email addresses and will be delivered accordingly.

NOTE: For most cellphone providers, you also can send an email message for SMS delivery by the provider. The complete list of providers is available here. Email messaging for SMS requires that you know the cellphone provider for your recipient while standard SMS messaging does not.

# In lieu of SMS number, email is also OK
8431234567 Doe John
mary@doe.com Doe Mary
8435551212@txt.att.net Mr T

In smsmsg.txt, enter the text message to be sent.

Once you have all three files configured, run the script: /root/smsblast.

NOTE: Google has tightened security for using plain-text Google Voice passwords as this application and the next one require. Before you begin, log into your server as root and issue the following command: gvoice. Try logging in with your Google Voice credentials including @gmail.com in your username. If the login fails, perform the following steps using a web browser after logging into the same Google account: (1) Perform the Google Voice Reset Procedure. (2) Enable Less Secure Apps using this Google tool. Then immediately try logging in to gvoice from the CLI again.

17. Voice-Activated SMS Messaging (dial SMS)

In addition to message blasting, you also can dial 767 from any extension and dictate an SMS message to send through your Google Voice account. When prompted for the destination, simply enter the 10-digit SMS number of the recipient. This new implementation of SMS Dictator requires both Google Voice and IBM STT credentials. Follow this tutorial to get started with the latest code.

18. SMS Messaging with VoIP.ms

Incredible PBX for Asterisk-GUI also supports SMS messaging through VoIP.ms if you have an account and an SMS-enabled DID. See the VoIP.ms wiki for setup info on the VoIP.ms side.

To install the VoIP.ms SMS scripts, follow these steps:

cd /root
mkdir sms-voip.ms
cd sms-voip.ms
wget http://incrediblepbx.com/voipms-SMS.tar.gz
tar zxvf voipms-SMS.tar.gz

Edit voipms-sms.php and insert your VoIP.ms number that supports SMS messaging (no spoofing allowed!):

$SMSsender="8005551212";

Edit class.voipms.php and insert your VoIP.ms API credentials:

    /*******************************************\
     *  VoIPms - API Credentials
    \*******************************************/
    var $api_username   = 'yourname@youremail.com';
    var $api_password   = 'yourpassword';

Send an SMS message through VoIP.ms with the following command where smsnumber is the 10-digit number of the SMS recipient and "sms message" is the text message surrounded by quotes:

/root/sms-voip.ms/voipms-sms.php smsnumber "sms message"

NOTE: VoIP.ms has indicated that sooner or later there will be a penny per message charge for SMS messages; however, as of today, they’re still free.

19. SIP URI Calling with Incredible PBX

With one line of dialplan code, you can add Speed Dials for free SIP URI calling worldwide. The dialplan code is stored in the [CallingRule_SIP_URI] context in extensions_custom.conf. Just clone one of the existing entries, designate an extension to dial to connect to the SIP URI, and enter the SIP URI for the destination. Numerous SIP providers support assignment of SIP URI’s to DIDs for unlimited free calling from anywhere in the world. Here’s a sample using a speed dial code of 53669 that connects you to SIP URI 2233435945@sip2sip.info: exten = 53669,1,Dial(SIP/2233435945@sip2sip.info)

20. IVR Demo of Incredible PBX Apps

The easiest way to try out a number of the Incredible PBX applications is to take the IVR Demo for a spin. Just pick up any phone and dial 3366 (D-E-M-O). The sample code for the IVR is available for review and modification in the IVR section of the GUI. There’s also a sample Stealth AutoAttendant. This plays a brief greeting and then rings an extension or ring group. During the greeting, you could configure the application to allow button presses to branch to other applications on your PBX, hence the Stealth name since the codes are not disclosed to callers.

21. Backup & Restore with Incredible PBX

Incredible Backup and Restore scripts are still under development for Incredible PBX 13-13 because of recent changes in Asterisk. If backups are important to you, we strongly recommend you consider a $2.50/month cloud server at Vultr using our referral code. For an additional 50 cents per month, you get weekly image backups of your server that can be restored with a couple of button clicks. It’s the cheapest insurance you can buy for your PBX!

22. AsteriDex – The Poor Man’s Rolodex

AsteriDex is a web-based phonebook application for Incredible PBX. You can access it from the main web menu. Scripts are also available to import your contacts from Outlook and Google Contacts.

23. Voice Dialing with AsteriDex (dial 411)

If you have voice recognition enabled on your server, you can call anyone in your AsteriDex database by dialing 411.

24. Speed Dialing with AsteriDex (dial 000+)

For those without voice recognition, Incredible PBX includes two speed dialing utilities. The first is accessed by dialing 412. Then enter any 3-digit dialcode from your AsteriDex database to complete the call. If you’d prefer to skip the intermediate step, dial 000 + the 3-digit speed dial code desired. The call will be placed immediately using your default outbound routes.

For a complete listing of your AsteriDex dial codes, execute this query:

mysql -u root -ppassw0rd asteridex -e "select name,dialcode from user1 order by name"

To automatically generate the 3-digit speed dial codes for everyone in your AsteriDex database using the first three letters of each name, run the following script from your web browser: http://your-server-ip/asteridex4/dialcode.php.

25. Telephone Reminders (dial 123)

Incredible PBX includes a sophisticated reminders system that lets you schedule individual or recurring reminders using your phone by dialing 123 or a web browser. A complete tutorial is available here. For phone reminders, a password is required to access the reminder system. Typically, these reminders set up a return call at a scheduled time that then plays back either a recorded message or a TTS message generated from the text you entered in the browser application. Incredible PBX also includes a new addition that lets you schedule web reminders that are delivered by email or SMS message.

26. DISA Access with Incredible PBX

Direct Inward System Access (aka DISA) is one of the great PBX inventions of the last 50 years. It’s also one of the most dangerous. It lets someone connect to your PBX and obtain dial tone to place an outbound call using your trunks… on your nickel. Typically, it is offered as an option with an IVR or AutoAttendant. The DISA extension is not preconfigured with Incredible PBX; however, you can easily set it up in the GUI by choosing Applications:DISA. Make up a very secure PIN before exposing DISA access to the outside world. It’s your phone bill.

27. Yahoo! News (Dial 951)

Yahoo! news headlines are available by dialing 951. The news option also is included in the sample IVR application.

28. Weather Forecasts by Phone (dial 947)

You can obtain a current weather forecast for most zip codes by dialing 947 (Z-I-P) and entering the 5-digit zip code.

29. ODBC Application Support for Asterisk

If you’ve recently logged into your server as root, Automatic Update #4 added ODBC/MySQL application support for Asterisk. You can try out a few sample applications that are included to get you started. Dial 222 and enter 12345 for the employee number. This retrieves an employee name from the MySQL timeclock database using Asterisk. Dial 223 to retrieve an AsteriDex name and phone number by entering the 3-character dialcode. You then have the option of placing the call by pressing 1. Once you have created accounts for Travelin’ Man 4, you can dial 864 (T-M-4) to WhiteList an IP address for that account after entering the account number and matching PIN. Use the * key for periods in the IP address.

30. Today in History (Dial T-O-D-A-Y)

It’s always interesting to find out what happened Today in History. And Incredible PBX now delivers it by phone. Just dial 86329 (T-O-D-A-Y) for a walk down memory lane.

31. Time of Day

Speaking of yesteryear, if you grew up dialing TI-4-1212 for the time of day, Ma Bell may have discontinued the service, but we haven’t. Now you can do it on your very own PBX.

If you want your users to be able to dial in for the time directly by dialing extension, here’s how. In the GUI, choose Admin:Custom Destinations:Add Destination. Set up a Time of Day description with a target of new-time,s,1 and save your entry. Now Enable an Application:Misc Application:Add Application with a Feature Code of 8463, Time of Day description, and point it to Custom Destination:Time of Day. Save your entry and then dial 8463 (T-I-M-E) for the Time of Day.

WebMin: The Linux Swiss Army Knife

There is no finer Linux application than WebMin. There is no more dangerous Linux application than WebMin. You’ve been warned. We heartily recommend WebMin as a tool to LOOK at your server’s settings. We strongly discourage changing anything in WebMin unless you totally know what you are doing. This is especially true with management of Linux applications that make up the core of Incredible PBX: the Linux kernel, SendMail, IPtables, Apache, MySQL, PHP, and…

To access WebMin, visit the following link with a web browser using the actual IP address of your server: https://ip-address:9001/. The username is root. The password is your root password. WebMin has root privileges to your server. Reread paragraph 1 and act accordingly.

For an exhaustive tutorial on WebMin, download The Book of WebMin by Joe Cooper. For a more recent commercial offering, take a look at Michal Karzyński’s WebMin Administrator’s Cookbook.

Enjoy your new Gotcha-Free PBX, and… Happy Cyber Monday! It’s always been one of the happiest days of the year around our office.

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

Originally published: Monday, November 27, 2017


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle 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 Forums. 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, ours 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.



NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!



Need help with Asterisk? Visit the PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Back to Basics: Configuring Extensions, Trunks & Routes

With last week’s release of Incredible PBX 13-13 Lean with Asterisk® 13 and FreePBX® 13 GPL modules, it seemed like an opportune time to revisit the initial setup process of an Asterisk-based PBX. Configuring extensions, trunks, and routes are the fundamental steps in successfully interconnecting your PBX to the telecommunications network. So today we’ll walk through the initial setup process in some detail for those that are just getting started. And we think old-timers may find some hidden nuggets in the exercise as well.

Overview of the Initial Asterisk Setup Process

For those new to PBXs, here’s a two paragraph summary of how Voice over IP (VoIP) works. Phones connected to your PBX are registered with Extensions so that they can make and receive calls. When a PBX user picks up a phone and dials a number, an Outbound Route tells the PBX which Trunk to use to place the call based upon established dialing rules. Unless the dialed number is a local extension, a Trunk registered with some service provider accepts the call, and the PBX sends the call to that provider. The provider then routes the call to its destination where the recipient’s phone rings to announce the incoming call. When the recipient picks up the phone, the conversation begins.

Looking at things from the other end, when a caller somewhere in the world wishes to reach you, the caller picks up a telephone and dials a number known as a DID that is assigned to you by a provider with whom you have established service. When the provider receives the call to your DID, it routes the call to your PBX based upon destination information you established with the provider. Your PBX receives the call with information identifying the DID of the call as well as the CallerID name and number of the caller. An Inbound Route on your PBX then determines where to send the call based upon that DID and CallerID information. Typically, a call is routed to an Extension, a group of Extensions known as a Ring Group, or an IVR or AutoAttendant giving the caller choices on routing the call to the desired destination. Once the call is routed to an Extension, the PBX rings the phone registered to that Extension. When you pick up the phone, the conversation begins.

Configuring Asterisk to Support NAT-Based Routing

With a VoIP server, many PBXs and Extensions are housed behind a NAT-based router that is found in most homes and businesses. These routers assign private IP addresses that are not accessible from the Internet. This causes SIP routing headaches because there are actually two legs to every call, one on the private IP address of your server or extension and another on the public Internet with an entirely different IP address. Routers supposedly handle this handoff of the call using Network Address Translation (NAT) and SIP ALG. With Asterisk-based PBXs, we want the PBX itself to handle the NAT chores so it is critically important to do three things when setting up your PBX. First, turn off SIP ALG on every router used by your PBX and every extension connected to your PBX. Second, tell your PBX about your public and private IP address setup. Step #2 is done in the Incredible PBX GUI with a browser. Login as admin and choose Settings:Asterisk SIP Settings. In the NAT Settings section of the form, click Detect Network Settings. Make sure your public and private IP addresses are correctly listed. Then click Submit and reload your dialplan when prompted. Failure to perform BOTH of these steps typically results in calls with one-way audio, i.e. where either you or the called party can’t hear the other party in the conversation. The third rule to remember is to always configure SIP Extensions on your PBX with NAT Mode=YES. This is rarely harmful and failure to configure SIP extensions in this way typically causes one-way audio in calls as well. IAX extensions avoid NAT issues.

Configuring Extensions with Incredible PBX GUI

Extensions are created using the Incredible PBX GUI: Applications:Extensions. Many SIP phones expect extensions to communicate on UDP port 5060. If this is the case with your SIP phone or softphone, then always create Chan_SIP extensions which communicate on UDP 5060. If your SIP phone or softphone provide port flexibility, then you have a choice in the type of SIP extension to create: Chan_SIP or the more versatile PJSIP. Just remember to always configure SIP extensions with NAT Mode=YES in the Advanced tab. If your VoIP phones or softphones support IAX connectivity, you may wish to consider IAX extensions which avoid NAT problems.

When you create a new Extension, a new entry is automatically created in the PBX Internal Directory. If you wish to allow individual users to manage their extensions or use the WebRTC softphone, then you will also have to create a (very) secure password for User Control Panel (UCP) access. Choose Admin:User Management and click on the key icon of the desired extension to assign a password for UCP and WebRTC access.

Configuring SIP Phones with Incredible PBX GUI

SIP phones and softphones typically require three pieces of information: the IP address of your server, the extension number, and the extension password. If you’re using a PJSIP extension, you also will need to change the port to UDP 5061. If your server is behind a NAT-based router, SIP phones also behind the same router need to use the private LAN address rather than the public IP address. If the SIP phones are outside the router protecting the PBX, then use the public IP address and make certain that you also map ports 5060 and 5061 from your router to the private LAN address of your PBX.

The PIAF Forum can provide you with helpful information in choosing high quality SIP phones. Yealink phones are highly recommended with minimal issues. Cisco phones are the most difficult to configure. Insofar as free softphones, we recommend the Zoiper 3 offerings for Windows, Mac, iOS, and Android. Zoiper 5 still is experiencing some growing pains. A key advantage of the Zoiper softphone is it supports IAX extensions which eliminate the NAT issues entirely. On the Mac platform, we also recommend the Telephone app which is available in the App Store. For SRTP communications, use Grandstream Wave.

Configuring Trunks with Incredible PBX GUI

Perhaps the most difficult component to configure in the PBX is the Trunk. Almost every provider has a different way of doing things. We’ve taken some of the torture out of the exercise by providing a script which will configure settings for dozens of providers in seconds. Once installed, all you need to do is edit the desired Trunk (Connectivity:Trunks), change the Disable Trunk entry to No, and insert your credentials in both the PEER Details and Registration string of the SIP Settings Outgoing and Incoming tabs.

To install the Trunk setups on your PBX, log into your server as root and issue the following commands only once:

cd /root
wget http://incrediblepbx.com/create-sample-trunks.tar.gz
tar zxvf create-sample-trunks.tar.gz
rm create-sample-trunks.tar.gz
./create-sample-trunks

Incredible PBX Wholesale Providers Access

Nerd Vittles has negotiated a special offer that gives you instant access to 300+ wholesale carriers around the globe. In lieu of paying the $650 annual fee for the service, a 13% wholesale surcharge is assessed to cover operational costs of TelecomsXchange. In addition, TelecomsXchange has generously offered to contribute a portion of the surcharge to support the Incredible PBX open source project. See this Nerd Vittles tutorial for installation instructions and signup details.

Configuring Google Voice with Incredible PBX GUI

The advantage of Google Voice trunks for those of you in the United States is that all of your calls within the U.S. and Canada are free. You can’t beat the price, and it has worked reliably for many, many years. There are three different ways to set up Google Voice trunks with Incredible PBX. For a one-time fee of $4.99 with this coupon, you can use the Simonics GV/SIP gateway to configure a Google Voice account using OAuth 2 authentication. Then just set up the Simonics SIP trunk on your PBX to point to the Simonics gateway. A second option is to choose the (recommended) OAuth 2 authentication method for Google Voice when you initially install Incredible PBX 13-13. Finally, you can choose plain-text passwords for Google Voice when you set up Incredible PBX. The drawback of this last option is Google has hinted that they may discontinue support of plain-text passwords.


Here are the initial setup steps on the Google side:

1. Set up a dedicated Gmail and Google Voice account to use exclusively for this Google Voice setup on your PBX. Head over to the Google Voice site and register. You’ll need to provide a U.S. phone number to verify your account by either text message or phone call.



2. Once you have verified your account by entering your verification code, you’ll get a welcome message from Mr. Google. Click Continue to Google Voice.



3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.



4. Once your phone number has been verified, choose a DID in the area code of your choice.



Special Note: Google continues to tighten up on obtaining more than one Google Voice number from the same computer or the same IP address. If this is a problem for you, here’s a workaround. From your smartphone, install the Google Voice app from iPhone App Store or Google’s Play Store. Then open the app and login to your new Google account. Choose your new Google Voice number when prompted and provide a cell number with SMS as your callback number for verification. Once the number is verified, log out of Google Voice. Do NOT make any calls. Now head back to your PC’s browser and login to https://voice.google.com. You will be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there’s still a way to get there. After you have set up your new phone number and opened the Google Voice interface, click on the 3 vertical dots in the left sidebar (it’s labeled More). When it opens, click Legacy Google Voice in the sidebar. That will return you to the old UI. Now click on the Gear icon (upper right) and choose Settings. Make sure the Google Chat option is selected and disable forwarding calls to whatever default phone number you set up.

5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller’s Number, and Global Spam Filtering:checked. The remaining entries should be blank.

6. Google Voice configuration is now complete. Sign out of your Google Voice account.


The Simonics GV-SIP Gateway Solution. Here’s the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we’ll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we’ll add Incoming and Outgoing Routes to tell Incredible PBX how to process Google Voice calls.

Now you’re ready to set up an account on the Simonics site. With this Nerd Vittles link, there’s a one-time fee of $4.99.

1. Start by registering your new Google account.

2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.

3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your PBX.



4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.

Now it’s time to configure your Simonics trunk in Incredible PBX. Start by logging into the web interface as admin with your admin password from above. Click Connectivity:Trunks and choose the Simonics trunk in the PBX Configuration menu. The Simonics trunk template will display:

1. Untick the Disable Trunk check box.

2. In Outbound CallerID, insert your 10-digit Google Voice number.

3. In username, insert GV1 followed by your 10-digit Google Voice number.

4. In secret, insert your Simonics SIP password.

5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)

6. In the Registration String after the colon, insert your Simonics SIP password.

7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.

8. Click Submit Changes and then Reload the Dialplan when prompted.


Configuring GV Trunk with Motif in the GUI. If you elect to configure your Google Voice trunk natively using the Incredible PBX GUI, you first will need to obtain a Refresh_Token if you elected to use OAuth 2 authentication.

1. Be sure you are still logged into your Google Voice account. If not, log back in at https://voice.google.com.

2. In a separate browser tab, go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.

3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).

  3a. Check the box: Use your own OAuth credentials
  3b. Enter Incredible PBX OAuth Client ID:

466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com

  3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
  3d. Click Close

4. Click Step 1: Select and Authorize APIs (as shown below)

  4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
  4b. Click Authorize APIs (blue) button.

5. Click Step 2: Exchange authorization code for tokens

  5a. Click Exchange authorization code for tokens (blue) button

  5b. When the tokens have been generated, Step 2 will close.

6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in the Incredible PBX GUI. Store this refresh_token in a safe place. Google doesn’t permanently store it!

7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.

Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.

Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don’t skip any. It’s easy once you get the hang of it. And you’ll sleep better.

Now you’re ready to configure your Google Voice account in Incredible PBX. You do it from within the Incredible PBX GUI by choosing Connectivity:Google Voice. Just plug in your Google Voice Username, enter your refresh_token from Step #6 above as your Google Voice Password, enter your 10-digit Google Voice Phone Number, and check the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit and Apply Settings to save your new entries.

If you elected to use plain-text passwords for Google Voice, simply skip obtaining OAuth 2 credentials and substitute your plain-text password for the refresh_token when you create the Google Voice trunk above. If you have trouble getting Google Voice to work using a plain-text password, 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.

IMPORTANT: Once you’ve entered your credentials, you MUST restart Asterisk from the Linux command line, or Google Voice calls will fail: amportal restart

Configuring Outbound Routes in Incredible PBX GUI

Outbound Routes serve a couple of purposes. First, they assure that calls placed by users of your PBX are routed out through an appropriate trunk to reach their destination in the least costly manner. Second, they serve as a security mechanism by either blocking or restricting certain calls by requiring a PIN to complete the calls. For example, if you only permit 10-digit calls and route all of those calls out through a Google Voice trunk, there is zero risk of running up an exorbitant phone bill because of unauthorized calls unless you’ve deposited a lot of money in your Google Voice account. This raises another important security tip. Never authorize recurring charges on credit cards registered with your VoIP providers and, if possible, place pricing limits on calls with your providers. If a bad guy were to break into your PBX, you don’t want to give the intruder a blank check to make unauthorized calls. And you certainly don’t want to join the $100,000 Phone Bill Club.

To create outbound routes in the Incredible PBX GUI, navigate to Connectivity:Outbound Routes and click Add Outbound Route. In the Route Settings tab, give the Outbound Route a name and choose one or more trunks to use for the outbound calls. In the Dial Patterns tab, specify the dial strings that must be matched to use this Outbound Route. NXXNXXXXXX would require only 10-digit numbers with the first and fourth digits being a number between 2 and 9. Note that Outbound Routes are searched from the top entry to the bottom until there is a match. Make certain that you order your routes correctly and then place test calls watching the Asterisk CLI to make sure the calls are routed as you intended.

Configuring Inbound Routes in Incredible PBX GUI

Inbound Routes, as the name implies, are used to direct incoming calls to a specific destination. That destination could be an extension, a ring group, an IVR or AutoAttendant, or even a conference or DISA extension to place outbound calls (hopefully with a very secure password). Inbound Routes can be identified by DID, CallerID number, or both. To create Inbound Routes, choose Connectivity:Inbound Routes and then click Add Inbound Route. Provide at least a Description for the route, a DID to be matched, and the Destination for the incoming calls that match. If you only want certain callers to be able to reach certain extensions, add a CallerID number to your matching criteria. You can add Call Recording and CallerID CNAM Lookups under the Other tab.

Bug Fix for Incredible Fax Installer

If you installed Incredible PBX 13-13 during the past week, be advised that removal of a GitHub repo will prevent the Incredible Fax installer in /root from completing successfully. Here’s the fix:

sed -i 's|joshnorth|wardmundy|' /root/incrediblefax11.sh

Also note that, with the Incredible PBX 13-13 Lean install, you must manually create a Custom Destination using the GUI. This is the destination to use for receipt of incoming faxes. The settings for the new Custom Destination should look like this:

target -> custom-fax-iaxmodem,s,1
label  -> Fax (HylaFax)
return -> no

You now have a functioning PBX. Down the road, we’ll tackle some of the more esoteric features of Asterisk so… Stay Tuned!

Published: Monday, October 30, 2017  


NEW YEAR’S TREAT: If you could use one or more free DIDs in the U.S. with unlimited inbound calls and unlimited simultaneous channels, then today’s your lucky day. TelecomsXChange and Bluebird Communications have a few hundred thousand DIDs to give away so you better hurry. You have your choice of DID locations including New York, New Jersey, California, Texas, and Iowa. The DIDs support Voice, Fax, Video, and even Text Messaging (by request). The only requirement at your end is a dedicated IP address for your VoIP server. Once you receive your welcome email with your number, be sure to whitelist the provider’s IP address in your firewall. For Incredible PBX servers, use add-ip to whitelist the UDP SIP port, 5060, using the IP address provided in your welcoming email.

Here’s the link to order your DIDs.

Your DID Trunk Setup in your favorite GUI should look like this:

Trunk Name: IPC
Peer Details:
type=friend
qualify=yes
host={IP address provided in welcome email}
context=from-trunk

Your Inbound Route should specify the 11-digit DID beginning with a 1. Enjoy!



Need help with Asterisk? Visit the PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Sneak Peek: Incredible PBX with FreePBX 13 GPL Modules


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As many of you know, the Asterisk® 13.16 and 13.17 releases caused serious breakage with FreePBX® 12 GPL deployments. These have been resolved in the new Asterisk 13.18 release candidate, but it prompted us to do a little exploring for alternatives while many of you were partying at AstriCon in Orlando last week. To give credit where credit is due, the Sangoma developers have made some impressive improvements with their FreePBX 13 release. And now we’re pleased to introduce a preview of the new Incredible PBX 13-13 platform featuring almost all of the FreePBX 13 GPL modules. For those that prefer lean and mean implementations of Asterisk, this preview edition is for you. You get the latest version of Asterisk compiled from source, and you get a base install of the FreePBX 13 GPL modules. The only preconfigured component in the build is the Travelin’ Man 3 firewall which locks down your server to trusted providers and IP addresses which you authorize.

We would hasten to add that Incredible PBX 13-13 is still a work in progress. Changes and improvements are released almost daily. If you don’t have that pioneering spirit, then you may wish to hold off for a few more weeks until all of the kinks have been worked out. Having said that, part of the fun of participating in the open source community is rolling up your sleeves and trying out new things. Offering your suggestions as we move along almost always yields a better end product both for you and the rest of the open source community. While we don’t ordinarily recommend deployment of preview editions for production systems, it’s worth noting that both Asterisk 13 and FreePBX 13 have a lot of miles on them and are generally regarded as the production-ready platforms for serious deployments.

So let’s get started. 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 such as Vultr. We recommend a minimum 1GB of RAM although you can get away with 512MB of RAM if you also create a swapfile. 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 or 7. The Incredible PBX installer will load all of the necessary components to support Asterisk and FreePBX as well as future Incredible PBX applications.

Installing a Base CentOS Operating System

Let’s begin by installing 64-bit CentOS 6.9 or 7 on your favorite hardware or Desktop. Or you may prefer to use a Cloud provider1 that already offers a preconfigured CentOS image. Two reasons we prefer Vultr are (1) Incredible PBX runs fine on their least costly $2.50/mo. platform and (2) for an extra 50 cents a month, you can add automatic backups to your server platform. In you’re using a Cloud platform, you can skip the rest of this section. Just choose CentOS 6 or 7 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 6.9 minimal ISO or the CentOS 7 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, here are the settings:

Type: Linux
Version: RedHat 64-bit
RAM: 1024MB
Default Drive Options with 10GB+ 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

Boot your server with the ISO, and 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 Preview

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. It’s a two-step process. First, the installer will bring your version of CentOS up to current specs and load the necessary packages to support Asterisk and FreePBX. The first stage setup takes about 10-15 minutes.

cd /root
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
./create-swapfile-DO
./IncrediblePBX-13-13.sh

When the base install finishes, your server will reboot. Simply log back in as root and run the installer a second time using an SSH terminal or Putty. You’ll be prompted whether to implement Google Voice plain text or OAuth 2 passwords. OAuth is strongly recommended. Make your selection, and the installer will work its magic. Come back in 15 minutes.

./IncrediblePBX-13-13.sh

Reboot one final time when the installer finishes the setup, and your server should be ready to go. Log back in as root. This will kick off the Automatic Update Utility to load any last minute additions, bug fixes, and security patches. After the status menu displays, run the following apps to set a very secure admin password for web access to the GUI and to choose your default time zone:

/root/admin-pw-change
/root/timezone-setup

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 server 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

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. Come join the discussion on the PIAF Forum and enjoy the ride!

Now Serving: The Incredible PBX 13-13 Whole Enchilada Upgrade

Continue Reading: Configuring Extensions, Trunks & Routes

Published: Monday, October 23, 2017  



Need help with Asterisk? Visit the PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

  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. []

The New Hybrid PBX: Why Settle for a One Trick Pony?


Let’s face it. It’s hard not to like the application development flexibility that Asterisk® offers, especially if you’re part of an organization that has very specific telephony needs. But the price you pay for "free" and putting all of your eggs in the Asterisk basket is painful. Here are a few of the hurdles that come to mind: security, NAT, one-way audio, remote users, CRM support, conferencing, painful upgrades to address frequent bug fixes, and, more generally, telephone management and support. We love Asterisk, but…

Most folks don’t buy all of their cars or groceries or computer software from a single company. So why do it with your phone system when you can take advantage of the best of all worlds, open source and commercial? To us, that’s the compelling case for integrating a 3CX commercial PBX into your Asterisk infrastructure. It’s a new iteration of what we used to call a hybrid PBX. And you can do it without cost for a full year to kick the 3CX tires and provide your mobile users with transparent phone service regardless of where they are roaming. Using the special Nerd Vittles signup link, you get a custom version of 3CX that supports 4 simultaneous calls, 10-user web meetings, unlimited trunks, and 10 or more extensions. After the first year, you can either spring for less than $100 a year to maintain the 3CX free PBX platform and mobile clients with pain-free updates, or you can upgrade to a more robust 3CX Pro commercial offering with a much expanded feature set including call center technology and seamless CRM integration with MS Exchange, Salesforce, Microsoft Dynamics, Microsoft Outlook, Office 365, Google Contacts, Exact Online, Freshdesk, Datev, Zendesk, Nutshell, vtiger, EBP, Insightly, amoCRM, Bitrix24 and Act. What’s not to like?

If you’re a frequent Nerd Vittles visitor, you already know that the 3CX clients for iOS, Android, Windows, and Macs are one of our favorite telephony apps of all time. The ease with which the 3CX client can be configured with a single click on an email attachment is revolutionary. And, once configured, the fact that you never again experience a NAT problem with a SIP call is nothing short of miraculous. As we’ve previously mentioned, the 3CX Client provides a nearly perfect mobile client for those that rely upon Asterisk. Now 3CX is poised to release an even easier configuration procedure for their mobile clients in update 2 for version 15.5. Simply log into your 3CX web client on a PC or Mac and choose the Settings:QR Code option from the menu bar. 3CX will present a QR code to activate the 3CX Client for your smartphone. Scan it using the 3CX Client app on your smartphone and, presto, your phone is instantly provisioned. It doesn’t get any easier than this…



Let’s spend a little time reviewing our favorite Hybrid PBX setup. In this scenario which is perfect for small businesses with a mobile workforce, the setup looks like this. An Asterisk server is deployed to manage company trunks including Google Voice, voicemail, IVRs, custom apps, and extensions for every employee. Then we add a 3CX free PBX, interconnect it with the Asterisk PBX, and assign a 3CX extension for every employee. The 3CX extensions will all tie back to the employee extensions on the Asterisk PBX. It obviously simplifies things if you keep your number schemes consistent. For example, extension 7000 on the Asterisk PBX could be matched to extension 000 on the 3CX PBX. Then we set up outbound trunks on both the Asterisk PBX and 3CX to dial a 9 prefix to reach extensions on the other PBX. So dialing 9000 on the Asterisk PBX would connect the caller to extension 000 on the 3CX PBX. On the 3CX side, dialing 9000 would connect the caller to extension 7000 on the Asterisk PBX in our example. And, of course, 3CX Clients can reach any number worldwide using Asterisk outbound trunks by dialing a 9 prefix and then the long distance number. Our previous tutorials will walk you through setting this up with Incredible PBX® 13, Issabel™, any FreePBX®-based PBX, or even Wazo. Once you complete the 5-minute setup, mobile users can take advantage of all the powerful features on any 3CX Client platform while still receiving their incoming calls from the Asterisk-based office PBX by simply forwarding their extension to their matching 9XXX destination on the 3CX platform. This will ring their 3CX Client anywhere in the world with nothing but a Wi-Fi connection! And it’s a free call.



Published: Monday, October 16, 2017  



Need help with Asterisk? Visit the PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Game Changer: Hooking Up Facebook with Incredible PBX

There aren’t many VoIP discoveries that get us this excited about the future of telecom. But merging with 1.5 billion users plus Facebook’s enormous talent pool and technology resources is definitely something worthy of your attention. What a Facebook marriage with the VoIP platform could mean for the future of telecommunications is nothing short of earth-shattering. Few people still have home phones. Almost everyone has a Facebook account and a cellphone. If VoIP solutions for businesses fail to take those last two sentences into account, commercial PBX’s days are numbered… and it’s not a big number.

So why integrate Facebook Messenger into your PBX? The screenshot above says it all.

Think of the possibilities. Using Facebook Messenger on your smartphone or desktop PC, you could query a CRM database running on your VoIP server and instantly connect to anyone in the world by making a free call or sending a free text message. Using Facebook Messenger, you or any designated employee could receive instant alerts when a new voicemail or fax arrived on your PBX. Using Facebook Messenger, the Call Center possibilities are virtually endless as documented here. Using Facebook Messenger, you as an administrator could literally manage your entire fleet of PBXs from the convenience of your smartphone… anywhere in the world. While the Facebook Messenger platform does not independently support phone calls between its users today, it’s just a matter of time. Look at the name of the product. Is there any doubt where this project is headed given the fact that Apple already supports free calling with Facetime, Microsoft supports free calling with Skype, Google supports free calling with Google Voice, and Amazon supports free calling with its Echo platform?

Facebook integration is revolutionary in another way as well. It heralds the arrival of chatbots to do the heavy lifting for telecom businesses as well as system administrators. Just as ATMs revolutionized banking, chatbots are poised to do much the same thing for communications and Internet support. Down the road, we’ll document how to take advantage of this chatbot technology using Facebook Messenger.

We need to learn to walk before we can run. So today we’ve developed a Facebook webhooks integration project for Incredible PBX® that is perfect for administrators, whether you manage a home PBX or a dozen PBXs for an organization. We’ll get to some of the other possibilities in future articles. Setting this up is the best way we can think of to get your creative juices flowing to consider what’s possible and to identify where to go next. When we’re finished, you’ll have a Facebook Messenger platform from which you can issue any Linux® or Asterisk® command to your server. And, you’ll be able to send messages from your PBX to Facebook Messenger to identify any events you wish to monitor, whether it’s phone calls, or voicemails, or receipt of faxes, or even VoIP provider outages. In addition, you can even reroute calls by entering simple call forwarding commands in Messenger.

Before we get started, let’s get all of the legal stuff out of the way up front. WE PROVIDE OPEN SOURCE, GPL CODE TO OUR READERS AT NO COST. ALWAYS HAVE. ALWAYS WILL. THE TRADEOFF IS YOU MUST AGREE TO ACCEPT ALL RISKS INHERENT IN USING THE SOFTWARE, WHETHER THOSE RISKS ARE KNOWN OR UNKNOWN TO YOU OR TO US. THE SOFTWARE IS PROVIDED "AS IS" AND MAY BE USED AS DELIVERED, OR YOU MAY MODIFY IT TO MEET YOUR OWN NEEDS SUBJECT TO THE TERMS OF THE GPL 2 LICENSE AVAILABLE HERE. IF YOU ARE UNWILLING TO AGREE TO THESE TERMS AND CONDITIONS, STOP READING HERE AND MOVE ON TO SOME OTHER WEB SITE. OTHERWISE, LET’S BEGIN WHAT WE PROMISE WILL BE A TERRIFIC ADVENTURE.

Overview of Facebook Messenger Webhooks Project

Here is a thumbnail sketch of what we’ll be covering today. Once you get an SSL certificate installed for your server, the remaining steps are a walk in the park. When we’re finished, you’ll have a Facebook Messenger platform that is seamlessly integrated with your PBX. The current software release supports Incredible PBX 13 with CentOS 6, Incredible PBX for Issabel, and Incredible PBX for Wazo. Minor tweaking required for other Asterisk platforms.

  • SSL Certificate – Obtaining and installing an SSL certificate for your web server
  • Security – Locking down your server for safe, secure Facebook Messenger access
  • Incredible PBX Webhooks App – Installing the server-side webhooks software
  • Facebook Integration – Interconnecting Facebook Messenger and Incredible PBX
  • Outbound Call Setup – Configuring Incredible PBX to make outbound calls from FB
  • Incoming Call Alerts – Configuring Incredible PBX for FB Messenger call alerts
  • Webhooks Feature Set – Our tutorial covering all supported webhook commands
  • SMS Messaging – Configuring Incredible PBX for SMS Messaging support with FB
  • Webhooks Tips & Tricks – Adjusting our code to meet your own requirements

Obtaining and Installing an SSL Certificate

Believe it or not, the hardest part of today’s project was covered in last week’s Nerd Vittles tutorial. It walked you through obtaining and installing an SSL Certificate on any of the major Incredible PBX platforms. This gets your server configured to use secure and encrypted web communications via HTTPS which is both a Facebook requirement and a smart idea. There’s no need to read further until you get your server working properly with an SSL certificate because the Facebook integration component will fail until you get HTTPS access squared away. So start there and return here when you’re finished.

The Most Important Piece of the Puzzle: SECURITY

If you’ve been following Nerd Vittles over the years, you already know that our most important consideration with any PBX deployment is security. A PBX without a secure firewall is an invitation for an astronomical phone bill. Today’s setup assumes you already have deployed Incredible PBX with its Travelin’ Man 3 firewall that provides a whitelist of IP addresses that may access (or even see) your server. By definition, Facebook Messenger is a public platform available to everyone in the world. So how do we safely integrate it into your PBX while preserving the security of your server and its telecom resources? We do it in several ways. First, Facebook Messenger Webhooks are tied to a commercial Facebook page even though you don’t need a business in order to create the page. As the owner of that Facebook Page, you have to authorize users to access the page. DON’T! Make this a page that is solely dedicated to managing your PBX through Messenger. DO NOT USE THIS FACEBOOK PAGE AS THE PUBLIC FACE FOR YOUR BUSINESS! Also make certain that your Facebook credentials include a very secure password… as if the integrity of your PBX depended upon it. IT DOES! So long as you follow these guidelines, Facebook’s own security mechanisms will protect your PBX from intrusion. If this discussion makes you nervous, our last topic today will show you how to remove components from the code to eliminate any functionality you wish to turn off.

As configured, Facebook Messenger Webhooks won’t work at all with Incredible PBX because the firewall should block all web access to your server. This requires a change on the Incredible PBX for Wazo platform which we will cover momentarily. The way we will provide Facebook access is by adding the Facebook server IP addresses to the existing whitelist, and then we’ll run a bash script every night to keep the Facebook IP addresses current.

In the past, we opened TCP port 443 (HTTPS) to public access on the firewall with Incredible PBX for Wazo. Instead, we relied upon web server authentication for access to the Wazo, Telephone Reminders, and AsteriDex services. That needs to be changed before you interconnect with Facebook Messenger, and we’ll include that in the commands to whitelist the Facebook servers below.

1. To secure port 443 in your firewall, be sure that the port is not exposed in /etc/sysconfig/iptables (CentOS) or /etc/iptables/rules.v4 (Debian/Ubuntu/Raspbian). And then restart the Incredible PBX firewall.

sed -i 's|443|450|' /etc/sysconfig/iptables
sed -i 's|443|450|' /etc/iptables/rules.v4
iptables-restart

2. Verify your new configuration: iptables -nL. Search for 443 and make certain it is NOT in the whitelist.

3. Verify that the whois package is installed on your server by issuing the command: whois. If you get a file not found error, install the package using the top line for CentOS and the bottom line for Debian/Ubuntu/Raspbian:

yum install whois
apt-get install whois

4a. For Issabel and Incredible PBX 13, add to the end of /usr/local/sbin/iptables-restart these lines to whitelist the FB servers. Then restart the firewall: iptables-restart

whois -h whois.radb.net -- '-i origin AS32934' | grep ^route: | sed "s|route:     |/usr/sbin/iptables -A INPUT -s |" | sed "s|$| -p tcp -m tcp --dport 443 -j ACCEPT|" > /usr/local/sbin/iptables-facebook
chmod +x /usr/local/sbin/iptables-facebook
/usr/local/sbin/iptables-facebook

4b. For Incredible PBX for Wazo, add to end of /usr/local/sbin/iptables-restart these lines to whitelist the FB servers. Then restart the firewall: iptables-restart

whois -h whois.radb.net -- '-i origin AS32934' | grep ^route: | sed "s|route:     |/sbin/iptables -A INPUT -s |" | sed "s|$| -p tcp -m tcp --dport 443 -j ACCEPT|" > /usr/local/sbin/iptables-facebook
chmod +x /usr/local/sbin/iptables-facebook
/usr/local/sbin/iptables-facebook

5. Verify your new configuration: iptables -nL. You should see numerous whitelist entries for port 443 at the end of the listing.

6. Add the following command at the bottom of /etc/crontab to assure that the Facebook server IP addresses are kept current:

20 0 * * * root /usr/local/sbin/iptables-restart >/dev/null 2>&1

7a. For Issabel and Incredible PBX 13, create new web directory, set ownership/permissions to house the Facebook Messenger webhooks, and add a sample web page:

mkdir /var/www/html/fb
echo "Hello World" > /var/www/html/fb/index2.php
chown -R asterisk:asterisk /var/www/html/fb

7b. For Incredible PBX for Wazo, create web directory, set ownership/permissions to house the Facebook Messenger webhooks, and add a sample web page:

mkdir /var/www/html/fb
echo "Hello World" > /var/www/html/fb/index2.php
chown -R asterisk:www-data /var/www/html/fb
chmod -R 775 /var/www/html/fb

8a. For Issabel and Incredible PBX 13, no further configuration is required.

8b. For Incredible PBX for Wazo, we need to enable access to the fb web directory. Edit /etc/nginx/locations/https-available/01_incrediblepbx:

At the top of the file, add the following:

location ~* ^/fb/. *\(?:ico|css|js|gif|jpe?g|png)${
 root /var/www/html;
}

At the bottom of the file, add the following:

location ~ /fb/ {
 root /var/www/html;
 index index.php;
 try_files $uri $uri/ =404;
 fastcgi_param SCRIPT_FILENAME $document_root$fastcgi_script_name;
 fasstcgi_index index.php;
 include fastcgi_params;
 fastcgi_pass unix:/var/run/php5-fpm.sock;
}

Finally, restart the NGINX web server: service nginx restart

9. Using a browser, verify access to sample page: https://SERVER-FQDN/fb/index2.php

Installing Incredible PBX Webhooks Application

Now it’s time to install the Incredible PBX webhooks application on your PBX:

cd /var/www/html/fb
wget http://incrediblepbx.com/incrediblewebhooks.tar.gz
tar zxvf incrediblewebhooks.tar.gz
rm incrediblewebhooks.tar.gz

For Issabel and Incredible PBX 13, adjust the file ownership and permissions like this:

chown -R asterisk:asterisk /var/www/html/fb
chmod -R 775 /var/www/html/fb

For Incredible PBX for Wazo, adjust the file ownership and permissions like this:

chown -R asterisk:www-data /var/www/html/fb
chmod -R 775 /var/www/html/fb

Hooking Up with Facebook

1. Visit the Facebook Developer’s Page and click Add a New App. Give your app a Display Name and provide your Contact Email. Match the letters in the box to get past the Security Check to display the Facebook Product List.

2. When the Facebook Product List appears, click Messenger and choose Setup.

3. In the Token Generation section, click Create a new Facebook Business Page to open a separate browser tab. Do NOT use a page that you use for other purposes! Company, Organization, or Institution is a good choice because there’s a Telecom Company category. Give your new page a Descriptive Name: incrediblepbx-podunk.

4. Return to your Token Generation browser tab and Select the Page you just created from the pull-down list (see Token Generation section of image below). Click Continue and OK to accept the default settings. Facebook then will generate a Page Access Token.

5. Copy the Page Access Token to your clipboard and paste it into the $access_token variable in the config.inc.php template in /var/www/html/fb. Write it down and keep it in a safe place. You’ll always need it to create new webhooks applications. This is the important link to talk to your Facebook Webhooks.

6. In the Webhooks section, click Setup Webhooks. In the Page Subscription form, enter the callback URL for your page. This is the https address to access your Facebook directory with a browser, e.g. https://YOUR-FQDN/fb. Make up a very secure Verify Token and enter it on the form and in the $verify_token variable in the config.inc.php template. This is the code Facebook will send to initially shake hands with your web page. The two entries must match to successfully set up your webhooks linkage. For Subscription Fields, check the Messages box. Then click Verify and Save. If it worked, you’ll get a Complete checkmark in the Webhooks section (see below). The last step is to again Select your Page in the Webhooks section to interconnect Facebook with your PBX. After choosing your page, be sure to click Subscribe or nothing will work. Here’s what a successful setup looks like:

7. To test things out, open Facebook Messenger on a desktop PC, Mac, or smartphone. Search Messenger for the Facebook page you linked to in the previous step. Then click on it to open it. Type howdy in the Message Box at the bottom of the dialog and click Send.

8. You should get an automated response that looks like this:

Hi there and welcome to BotWorld. SenderID:  13824822489535983

9. Copy the SenderID and paste it into cli-message.php together with Page Access Token from step #5, above.

Outbound Call Setup for Facebook Messenger

Outbound calling with Facebook Messenger works like this. You can connect to a specific number using the dial command. Or you can use the call command to look up an entry in your AsteriDex database. Messenger then will display the matching phone number and give you the option of placing the call. When the call is initiated, Incredible PBX will first call your designated CALL-PICKUP-NUMBER. It could be an extension or ring group of your choice. You could even specify a mobile phone number as the pickup destination provided your PBX supports at least two simultaneous outbound calls. Google Voice and many SIP providers can handle this with a single DID. Our personal preference is to route the pickup call to a trunk on a 3CX server which then sends the call to every 3CX client registered with the 3CX server. No NAT issues ever! Once you pick up the call on your designated phone, Incredible PBX will place the second call to the number you requested in Facebook Messenger. The two calls then are connected as if you had placed the call directly. The brief video below demonstrates how this works and the flexibility of using Acer’s $250 Chromebook Flip with Messenger and a 3CX client as a (free) WiFi-based web communications platform with Google Voice. It lets you place and take calls from anywhere in the world so long as you have Wi-Fi access. It’s a dirt cheap travel companion.




To make all of this work, you need to designate a phone in /var/www/html/fb/.cli-call to take outbound calls initiated from Facebook Messenger. This is either an extension number or a 10-digit CALL-PICKUP-NUMBER in the examples below. To set this up, edit .cli-call and choose one of the following examples. Comment out the other Channel options.

For Issabel and Incredible PBX 13, choose from the following:

#echo "Channel: SIP/701" > /tmp/cli.call
#echo "Channel: SIP/vitel-outbound/1CALL-PICKUP-NUMBER" > /tmp/cli-call
echo "Channel: Motif/gSOME-GV-NAMEgmailcom/1CALL-PICKUP-NUMBER@voice.google.com" > /tmp/cli.call

For Incredible PBX for Wazo, choose from the following:

echo "Channel: Local/701@default" > /tmp/cli.call
#echo "Channel: Local/CALL-PICKUP-NUMBER@default" > /tmp/cli.call

Incoming Call Alerts with Facebook Messenger

If you’ve always wished for screenpops to announce your incoming calls, you’re going to drool at the FB Messenger Webhooks implementation with Incredible PBX. It works (simultaneously) on desktop PCs, Macs, iPhones/iPads, Android devices, and Apple Watch:

To set up incoming call alerts with Facebook Messenger, just issue the commands for your platform as outlined below.

For Incredible PBX 13, add the following to the end of extensions_override_freepbx.conf in /etc/asterisk directory. Then reload Asterisk dialplan: asterisk -rx "dialplan reload"

[cidlookup]
include => cidlookup-custom
exten => cidlookup_1,1,Set(CURLOPT(httptimeout)=7)
exten => cidlookup_1,n,Set(CALLERID(name)=${CURL(https://api.opencnam.com/v2/phone/${CALLERID(num)}?format=pbx&ref=freepbx)})
exten => cidlookup_1,n,Set(current_hour=${STRFTIME(,,%Y-%m-%d %H)})
exten => cidlookup_1,n,Set(last_query_hour=${DB(cidlookup/opencnam_last_query_hour)})
exten => cidlookup_1,n,Set(total_hourly_queries=${DB(cidlookup/opencnam_total_hourly_queries)})
exten => cidlookup_1,n,ExecIf($["${last_query_hour}" != "${current_hour}"]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_1,n,ExecIf($["${total_hourly_queries}" = ""]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_1,n,Set(DB(cidlookup/opencnam_total_hourly_queries)=${MATH(${DB(cidlookup/opencnam_total_hourly_queries)}+1,i)})
exten => cidlookup_1,n,ExecIf($[${DB(cidlookup/opencnam_total_hourly_queries)} >= 60]?System(${ASTVARLIBDIR}/bin/opencnam-alert.php))
exten => cidlookup_1,n,Set(DB(cidlookup/opencnam_last_query_hour)=${current_hour})
exten => cidlookup_1,n,System(/usr/bin/php /var/www/html/fb/cli-message.php "Incoming call: ${CALLERID(number)} - ${CALLERID(name)}.")
exten => cidlookup_1,n,Return()

exten => cidlookup_return,1,ExecIf($["${DB(cidname/${CALLERID(num)})}" != ""]?Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}))
exten => cidlookup_return,n,Return()

;--== end of [cidlookup] ==--;

For Incredible PBX for Issabel, add this to the end of extensions_override_issabel.conf in /etc/asterisk directory. Then reload Asterisk dialplan: asterisk -rx "dialplan reload"

[cidlookup]
include => cidlookup-custom
exten => cidlookup_5,1,Set(CURLOPT(httptimeout)=7)
exten => cidlookup_5,n,Set(CALLERID(name)=${CURL(https://api.opencnam.com/v2/phone/${CALLERID(num)}?format=pbx&ref=issabelpbx)})
exten => cidlookup_5,n,Set(current_hour=${STRFTIME(,,%Y-%m-%d %H)})
exten => cidlookup_5,n,Set(last_query_hour=${DB(cidlookup/opencnam_last_query_hour)})
exten => cidlookup_5,n,Set(total_hourly_queries=${DB(cidlookup/opencnam_total_hourly_queries)})
exten => cidlookup_5,n,ExecIf($["${last_query_hour}" != "${current_hour}"]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_5,n,ExecIf($["${total_hourly_queries}" = ""]?Set(DB(cidlookup/opencnam_total_hourly_queries)=0))
exten => cidlookup_5,n,Set(DB(cidlookup/opencnam_total_hourly_queries)=${MATH(${DB(cidlookup/opencnam_total_hourly_queries)}+1,i)})
exten => cidlookup_5,n,ExecIf($[${DB(cidlookup/opencnam_total_hourly_queries)} >= 60]?System(${ASTVARLIBDIR}/bin/opencnam-alert.php))
exten => cidlookup_5,n,Set(DB(cidlookup/opencnam_last_query_hour)=${current_hour})
exten => cidlookup_5,n,System(/usr/bin/php /var/www/html/fb/cli-message.php "Incoming call: ${CALLERID(number)} - ${CALLERID(name)}.")
exten => cidlookup_5,n,Return()

exten => cidlookup_return,1,ExecIf($["${DB(cidname/${CALLERID(num)})}" != ""]?Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}))
exten => cidlookup_return,n,Return()

;--== end of [cidlookup] ==--;

For Incredible PBX for Wazo, edit /etc/asterisk/extensions_extra.d/cid-superfecta.conf. In the [xivo-subrgbl-did] context just below the n(keepon),Gosub(cid-superfecta,s,1) line, insert the following. Then reload the Asterisk dialplan: asterisk -rx "dialplan reload"

same = n,System(/usr/bin/php /var/www/html/fb/cli-message.php "Incoming call: ${XIVO_SRCNUM} - ${CALLERID(name)}.")

Incredible PBX Webhooks Feature Set

Now that we’ve got all the pieces in place and properly configured, let’s briefly walk through the various options that are available. With all commands, you use Facebook Messenger with your designated web page on any platform supported by Messenger.

dial 8005551212 – connects to designated extension and then calls 8005551212
call Delta – looks up Delta in AsteriDex and provides button to place the call
lookup Delta – looks up Delta in AsteriDex and provides button to place the call
!command – executes a Linux command, e.g. !asterisk -rx "sip show registry"
howdy – returns greeting and SENDER ID of your FB page (Hookup, item #9)
help – provides links to phone help as well as PIAF and Asterisk forums
sms 10-digit-SMS-number "Some message" – sends SMS message through GV
update – updates Messenger platform for Incredible PBX to the latest & greatest
anything else – returns whatever you typed as a response (for now)

Configuring Incredible PBX for SMS Messaging

We’ve implemented a traditional SMS messaging function in this build that let’s you send an SMS message to any phone if you have a Google Voice account and assuming you have pygooglevoice functioning properly on your PBX. The Google Voice account need not be registered as a trunk on the PBX. To use the feature, insert your Google Voice credentials including your plain-text password for a working Google Voice account in /var/www/html/fb/.smssend. Then test the SMS functionality by issuing the following command from the Linux CLI:

/var/www/html/fb/.smssend 10-DIGIT-SMS-NUMBER "Hello SMS World"

If an error occurs, the script will tell you what to try to fix it. Begin by Enabling Less Secure Apps. Then follow this link to relax Google Voice security on your account. If it still fails after trying both of these methods, you may have an old build of pygooglevoice. Here are the commands to bring your system up to current specs. Then try again.

cd /root
rm -r pygooglevoice
git clone https://github.com/wardmundy/pygooglevoice.git
cd pygooglevoice
python setup.py install
cp -p bin/gvoice /usr/bin/.

Once you’ve sent an SMS message successfully using .smssend, you can start sending SMS messages from within Messenger. Syntax: sms 10-digit-SMS-number "Some message"

Incredible PBX Webhooks Tips & Tricks

There’s lots to learn with Facebook Messenger Webhooks. When we started two weeks ago, there were no PHP resources on the web that offered much help. Lucky for you, our pain is your gain. The meat of the coconut is primarily stored in the index.php in your fb directory. Print it out and it will tell you everything you ever wanted to know about coding webhooks with PHP.

Disabling Shell Access. While shell access only provides asterisk or www-data permissions depending upon your platform, we’ve nevertheless heard from more than one source exclaiming what a dumb idea it is to put a webhooks shell command out in the wild. We trust our readers to use it responsibly and to always place it behind a firewall with public access to TCP port 443 blocked. If that design and the Facebook security mechanisms still leave you queasy, the short answer is to remove that block of code on your server or change the access code from ! to something much more obscure, e.g. YuKFoo!. This is easy to do but just be aware that if you change the access code or even remove the block of code, running the update command to load the latest release from Incredible PBX Headquarters will overwrite your changes. So it’s probably a better idea to rename the update command (line 248) as well so you don’t accidentally run it. You’ll find the shell command block of code beginning at line 64 in the 170928 version. If you change the access code to a different string, remember to change the substring "1″ reference in that line and the subsequent line to the actual length of your access code, e.g. YukFoo! is seven characters long so the number 1 would be replaced with 7 in BOTH lines 64 and 65.

Other Security Measures. We don’t trust anybody (and that includes Facebook) when it comes to accessing resources from our paid VoIP providers. We would encourage you to run this application on a dedicated Incredible PBX in the Cloud server that has only a single Google Voice trunk with no funds balance in that particular Google account. In this way, if your server is compromised, the worst thing that can happen is your Google account gets compromised or some stranger makes U.S. and Canadian calls without financial cost to you. Now that Cloud servers are available for less than $2 a month, it makes good sense to separate out applications that pose heightened security issues for you and yours. If you do decide to use a SIP provider rather than a Google Voice trunk, we strongly recommend restricting international calls and keeping a minimal balance in your account with no automatic replenishment enabled.

Getting Rid of Lenny. The help command included in the feature set provided is more of a traditional web page with buttons simulating hot links. We’ve included a nifty telephone option in the help features. It let’s you embed a phone number that is called using client-side integration whenever help is entered and the "Talk to Lenny" option is clicked:

What client-side integration means is the calls use any dialer available on the Messenger client’s platform. They are not sent to your PBX for processing. On a Mac or iPhone, Facetime provides free calls. On Windows, Skype provides paid calls. On Android devices, the Google Hangouts Dialer provides free calls. Facebook basically passes tel: +18005551212 to the client’s browser, and it’s up to the client’s browser to figure out how to process the call. We currently have the feature configured to "Talk to Lenny," but you could change it to Phone Home or Call the Office and enter your own phone number. Here are the commands to do it. Just replace "Phone Home" in the first command below with whatever label desired. Replace "8005551212″ in the second line with the number to be called. Leave the other Lenny entry and phone number as they are since they will be overwritten by these two commands. As noted above, your modifications will be overwritten whenever you execute the update command.

sed -i 's|Talk to Lenny|Phone Home|' /var/www/html/fb/index.php
sed -i 's|8436060444|8005551212|' /var/www/html/fb/index.php

Enhanced Calling Option. Beginning with the October 1 update which you can obtain by entering the update command in Messenger, you now have two calling options on some smartphone platforms. The call command still triggers an AsteriDex lookup on your PBX. But now you have a choice in how to place the call. (1) You can click the dial button to place the outbound call through your PBX, or (2) you can click on the retrieved phone number link to place the outbound call using the client-side resource available on your Messenger platform, e.g. Facetime, Skype, or Google Hangouts. In some circumstances, the client-side call may be preferable since it avoids the two-step calling procedure used by Asterisk. The choice is yours and may depend upon the availability and cost of the client-side call when placed from your calling location.

Special Thanks. Our special hat tip to Scott T. Tabor (@ABSGINC) for his pioneering work on Facebook Webhooks. You can visit the PIAF Forum and Scott’s blog to review how far we have come in just two weeks. Thanks, Scott.

Published: Monday, October 2, 2017  



Need help with Asterisk? Visit the PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Another Perfect Pair: Flawless VoIP with Wazo and 3CX


We previously documented how to interconnect an Issabel PBX with 3CX to take advantage of the best of both worlds. Today, we’ll again use the Nerd Vittles free 3CX server offering and interconnect it with a Wazo PBX. An added benefit of using Wazo is the fact that you can set up redundant (and free) HA servers with Wazo in minutes. Once we get the pieces in place, from Wazo extensions, you’ll be able to call your 3CX Clients by dialing 4 digits. And, from 3CX Clients, you can call Wazo extensions as well as all of your Asterisk® applications in the same way with the added bonus of being able to make outbound calls through your Wazo trunks by dialing any number with an 8 prefix from 3CX extensions. Once you have both of your PBXs running, the setup time to interconnect them is under 5 minutes.

Why would you want to maintain two PBXs? As we previously noted, the simple answer is the added flexibility you achieve coupled with a 99% reduction in VoIP headaches. If you haven’t yet used 3CX Clients on a PC or Mac desktop or on an iOS or Android device, you have missed perhaps the greatest VoIP advancement of the last decade. As the name suggests 3CX Clients connect to a 3CX server with less than a one-minute setup. They work flawlessly from anywhere using WiFi or cellular. Every function you’re accustomed to on a top-of-the-line desktop SIP phone works exactly the same on the 3CX clients: phonebook, hold, transfer, voicemail, chat, conferencing, and WebMeeting. It’s what every Unified Communications system should deliver. The silver lining is you can kiss all of your Asterisk NAT woes goodbye! If you ever travel or if you need remote phone access to your PBX infrastructure, you owe it to yourself to try a 3CX Client. We promise. You’ll never go back!



Building Your Wazo and 3CX Server Platforms

The prerequisite for interconnecting Wazo and 3CX servers is, of course, to install the two PBXs on platforms of your choice. Our preference is cloud-based servers because it avoids many of the stumbling blocks with NAT-based routers. If you know what you’re doing, you obviously can deploy the PBXs in any way you like. For the Wazo PBX, start with our latest Wazo tutorial. For 3CX, start with our introductory tutorial which includes a link to obtain a free perpetual license supporting 4 simultaneous calls and unlimited trunks. Then secure your server by adding the Travelin’ Man 3 firewall for 3CX. Once both servers are up and running, whitelist the IP address or FQDN of the Wazo PBX on the 3CX server and vice versa. You’ll find the add-ip and add-fqdn utilities in /root of each server.

Overview of Interconnection Methodology

If you’re new to all of this, suffice it to say that 3CX is a powerful, commercial PBX while Wazo provides a robust Asterisk RealTime implementation for basic telephony operation. The two systems are quite different in terms of their approaches to interconnectivity. While you can transparently interconnect one 3CX server to another one, you cannot accomplish the same thing when the second PBX is Asterisk-based. Instead, Wazo is configured as a SIP trunk on the 3CX platform. The limitation this causes is that extensions on the Wazo PBX can only direct dial extensions on the 3CX platform. Wazo-based extensions cannot utilize 3CX trunks to place outbound calls. There’s more flexibility on the 3CX side of things. 3CX extensions can place direct calls to Wazo extensions. They also can take advantage of Wazo’s trunks to place outbound calls. Additionally, as we noted above, 3CX extensions can take advantage of every Asterisk application hosted on the Wazo platform including all of the Incredible PBX® enhancements. This actually works out perfectly because you can deploy 3CX Clients for your end-users, and they can take advantage of all the extension and trunk resources on both the 3CX and Wazo platforms. It also greatly simplifies remote deployment by removing NAT one-way audio hassles while allowing almost instantaneous setup of remote 3CX Clients, even by end-users.

For our setup today, we’re assuming you have elected to use 3-digit extensions on both the Wazo and 3CX platforms. To call extensions connected directly to the alternate server, we will simply dial 8 + the extension number on the remote PBX. To make external calls from 3CX extensions using Wazo trunks, we will dial 8 + a 10-digit number. For international users, you can adjust the dialplan on both PBXs accordingly.

By default, SIP trunks are associated with a DID on the 3CX platform. We will register the 3CX DID trunk with Wazo to maintain connectivity; however, we will not register the corresponding trunk on the Wazo side with the 3CX server. Keep in mind that you can only route a 3CX DID to a single destination, i.e. an extension, a ring group, or an IVR. But we can use 3CX’s CallerID routing feature to send calls to specific 3CX extensions from Wazo extensions even using a single 3CX trunk. For each 3CX extension, we’ll create an Outbound Route on the Wazo side with a CallerID number that matches the 3CX extension number we wish to reach. On the 3CX side, we’ll create an Inbound CID Rule that specifies the extension number to which each matching CallerID number should be routed. This sounds harder than it actually is. So keep reading, and it’ll all make sense momentarily. Once you’ve set all of this up, we think you’ll agree that it makes sense to create the bulk of your extensions exclusively on the 3CX side.

Configuring Wazo for Interconnection to 3CX

Let’s begin by creating a Trunk on the Wazo side to connect to your 3CX server. In the Wazo GUI, choose IPBX:Trunk Management:SIP Protocol and + Add SIP Trunk.

In the General tab, fill in the blanks as shown below. Make up a very secure Password:

In the Signalling tab, fill in the blanks identified by arrows as shown below:

In the Advanced tab, fill in the blanks as shown below. Then SAVE the trunk settings.

Because we set up the Wazo trunk with a Default destination context, we don’t need an Incoming Route for the 3CX calls since they will be processed exactly as if they were dialed from a local extension on the Wazo PBX, i.e. local calls will be routed to extensions and outgoing calls through trunks will be routed using your existing Outbound Routes.

Finally, we need to create the Outbound Routes for calls originating from Wazo extensions that should be directed to specific extensions on the 3CX platform. You’ll need a list of the 3CX extension numbers you wish to enable on the Wazo platform, and we’ll need to create a separate Outbound Route for each 3CX extension to be enabled. Create the Outbound Routes using the template below after accessing Call Management:Outgoing Calls:+ Add Route.

In the General tab, we recommend including the 3CX extension in the Name field. The Context should be Outcalls, and the Trunk should be the 3CX001 trunk we created above.

In the Exten tab, specify the dialing prefix (9) followed by the 3CX extension number in the Exten field. Then choose 1 in the Stripnum field to tell Wazo to strip off the dialing prefix before sending the call to the 3CX PBX. Click SAVE to save your new outbound route settings. Repeat for each 3CX extension that should be accessible from the Wazo PBX.

Configuring 3CX for Interconnection to Issabel PBX

Now we’re ready to set up the 3CX side to interconnect with your Wazo PBX. Start by creating a SIP Trunk and fill out the template as shown below using one of the phone numbers associated with your Wazo PBX as the Main Trunk No.



Fill in the Trunk Details using the example below. Be sure to specify the actual IP address or FQDN of your Wazo server as well as the SIP credentials of 3CX for username and the actual password you set up on the Wazo side of things. The Main Trunk No will be the same as you entered in the previous step. Choose a Default Destination for the Trunk.

When the SIP Trunks listing redisplays, highlight your new Asterisk trunk and click Refresh Registration. The icon beside the Trunk should turn green. If not, be sure your IP address and password match the settings on the Wazo side. Remember to also whitelist the IP address of your 3CX server on the Wazo PBX using /root/add-ip and do the same for the Wazo PBX on the 3CX side. Don’t proceed until you get a green light!

Now we need two Outbound Routes for calls placed from 3CX extensions. One will handle calls destined for Local Extensions on the Wazo side. Our design is to place calls to Wazo extensions by dialing 8 + the 3-digit extension number. Adjust this to meet your own requirements. Be sure to set the Route as Wazo with a value of 1 for Strip Digits.

The other Outbound Route will handle calls destined for external calling with a Wazo trunk using a similar methodology. 3CX users will dial 8 + 10-digit number for calls to be processed by Trunks on the Wazo server.

Finally, we need an Inbound Rule for every 3CX extension that you wish to enable for remote calling from Wazo extensions. Use the Add CID Rule option to create each Inbound Rule using the sample below. In our example, we’re authorizing incoming calls to 3CX extension 003 where the CallerID number of the incoming call is 003. This template is exactly the same as what we used with the 3CX-Issabel setup previously.



Test Drive Your Interconnected Servers

Now we’re ready to try things out. From an extension on the 3CX server, dial 8 plus any 3-digit extension that exists on the Wazo server. Next, dial 8 plus a 10-digit number such as your smartphone. The call should be routed out of your Wazo server using the Trunk associated with the NXXNXXXXXX rule in your Wazo Outbound Routes. Finally, from an extension on your Wazo PBX, dial 9 plus 000 which should route the call to extension 000 on your 3CX server. Enjoy!

Published: Tuesday, September 5, 2017  


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 PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Finding the Perfect Phone Solution for Small Organizations

Many of us wear several hats during our business careers. One of those invariably is managing a community organization of some flavor. We frequently are asked for advice on what the ideal telephony solution would be for such an organization. The reason for the inquiries typically is because the Bell Sisters have now jacked up the cost of a single, business phone line to well over $100 a month. And that gets you local calls only unless you sign up for exorbitant additional charges for long distance calling. It’s worth noting that most of the individuals making these inquiries stress that they do not want to get in the business of managing a phone system. They’re looking for a plug-and-play, set-it-and-forget-it setup that will require minimal tweaking. My first question is always: "What’s your budget?" Then we explore (1) how many phones, (2) the frequency of calls, (3) the number of simultaneous calls, (4) the mix of local and long distance calling, and, last but not least, (5) the must-have feature set. No shocker: the budget is always near zero.




Today, we’re going to start on the bottom rung and work our way up the technology ladder. If you never thought smartphones and cellular would be part of this equation, guess again. $60 will now buy you a 4G LTE smartphone at WalMart, and monthly plans with unlimited calling in the U.S. start at $25 for Walmart’s Family Mobile plan, a far cry from the Ma Bell business phone rates. And you can keep your number! If you need multiple phones but only a single line, that’s not a problem either. Add a Link2Cell digital cordless phone system from Panasonic and now you have as many as 5 phones that can make and receive calls using your cellular connection via Bluetooth®. Some even support a second cellphone connection. With many you can build a phonebook on your cellphone and import it into all of your cordless phones. And, of course, voicemail is included as part of your cell plan. For those with poor cellular service, the Family Calling Plan supports free WiFi calling on many cellphones. And $10 extra buys you rollover international calling funds with 5¢/min. rates to Canada and Mexico. Calling rates to other countries are less than impressive and do not compare favorably with typical VoIP rates.

Cellular phone service isn’t for everyone, and there are considerably more choices in the Land of VoIP. The wrinkle with all of the VoIP solutions is that now you need internet service at the site of your organization. To say there is minimal competition in the internet service provider market is an understatement. If you’re lucky, you’ll have a choice between AT&T and one of the cable companies: Comcast, Charter, or Time Warner/Spectrum. The downside is it adds an additional $25 to $75+ to your monthly costs unless the organization already has Internet service that is used for purposes other than telephony. What won’t work for VoIP is satellite internet service because of latency issues.




Once you’re over the internet service hurdle, there are numerous VoIP choices for phone service depending upon your skillset. Again, let’s start on the bottom rung. If you can make it with one phone and one call at a time, it’s hard to beat Ooma Telo. $100 buys you a device that delivers landline-like phone service at a monthly cost of $4 (you only pay communications taxes and fees) to $10 depending upon the feature set you choose. The basic, fees-only plan gets you toll-free nationwide calling in the U.S., call waiting, caller ID, 911 service, a call log history and voicemail through Ooma’s online dashboard. The premium $10 a month plan adds a second line, free calling to Canada and Mexico, voicemail via email, call screening, do not disturb and call forwarding to an Android phone or iPhone. As with cellular service, you can keep your existing phone number. If you need WiFi connectivity or cellphone Bluetooth connectivity for your Ooma device, add $50. Otherwise, just plug a standard telephone into the Ooma hardware, and you’re good to go. You also could use a wireless phone system such as the ones described in the previous section to add up to five extensions.



If you need additional lines or phones, the $200 Ooma Office offering is worth considering. You can add as many users as desired for $19.95/month/each with every user getting unlimited U.S./Canada calling, CallerID service, and an impressive collection of business phone features (shown above). The cost of the VoIP phones for each user are not included. While the monthly service charges are pricey, you’re paying for the simplicity of never having to deal with the intricacies of configuring and managing a business phone system. However, you do have to purchase and configure a SIP phone for each user.



When you get beyond the single user, single line requirement, the sky opens up in the VoIP market. The savings go from getting part of your hundred dollars back each month to saving several hundred or thousands of dollars every month. What becomes important is how much of the deployment work you’re willing to undertake yourself. If the answer is not much, then the phone systems from one of our corporate sponsors, 3CX or RentPBX, are probably your best bets. Both offer turnkey VoIP solutions, and 3CX also has a worldwide dealer network to handle all of the deployment chores for you as well. While the front end costs with the 3CX commercial solution must be considered, the long-term savings more than cover these costs in your first year.

If you’re capable of making your own dinner by reading the directions off the side of a box, then you can probably handle many VoIP deployments yourself. The list of tasks goes something like this. You’ll either need a computer or cloud provider for a computing platform. Then you need a Linux operating system for that platform. Next, you need VoIP software to serve as your PBX. Services such as RentPBX handle setup of all three of these tasks for a monthly cost of $15. Or you can do it yourself and reduce the cost to $5 or less per month. We have dozens of tutorials to show you how.

At this juncture, you’re pretty much on your own except for our tutorials. The remaining tasks include purchasing and configuring phones for your users and configuring trunks from one or more VoIP providers, the folks that interconnect your phone calls to the people you are calling. Then you configure your PBX to route calls in and out of your PBX, and you’re in business. All of these tasks are managed using web-based GUI software, and there are plenty of tutorials to hold your hand every step of the way.

We’ll finish up today by walking you through one of our favorite open source VOIP setups. It provides free calling and faxing in the United States. Typical setup takes less than an hour, and the monthly cost is $3 which includes nightly backups of your entire PBX. These backups can be restored with a single button click.

FULL DISCLOSURE: 3CX, RentPBX, Amazon, Vitelity, and Vultr all provide financial support to Nerd Vittles and our open source projects. We’ve chosen these providers not the other way around. Our decisions were based upon their corporate reputation and the quality of their offerings and their pricing,

The Vultr/VoIP Open Source Solution

Begin by setting up an account at Vultr using our referral link. Then create a new instance choosing the smallest Server Size and CentOS 7/64-bit as the Server Type. Pick a Server Location that supports the $2.50 server size. Currently, Miami and New York are available. Once your virtual machine is running, you can activate automatic backups under the Server Information:Backups tab in the Vultr Control Panel.

(1) Once you’ve built and started your new virtual machine, log into your server as root using SSH/Putty and immediately change your root password: passwd.

(2) With the $2.50 size VULTR virtual machine, you must create a swapfile before proceeding. Here are the commands:

dd if=/dev/zero of=/swapfile bs=1024 count=1024k
chown root:root /swapfile
chmod 0600 /swapfile
mkswap /swapfile
swapon /swapfile
echo "/swapfile swap swap defaults 0 0">>/etc/fstab
sysctl vm.swappiness=10
echo vm.swappiness=10>>/etc/sysctl.conf
free -h
cat /proc/sys/vm/swappiness

(3) Now you’re ready to kick off the Issabel 4 install. Here are the commands:

cd /root
yum -y install wget nano dialog
wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash

When prompted for a MySQL password, use: passw0rd (with a zero). Choose a secure Issabel admin password for the GUI.

(4) After the reboot, log back in as root and install Incredible PBX for Issabel:

cd /root
wget http://incrediblepbx.com/IncrediblePBX11-Issabel4.sh
chmod +x IncrediblePBX11-Issabel4.sh
./IncrediblePBX11-Issabel4.sh

When prompted for a MySQL password, use: passw0rd (with a zero). Choose a secure Issabel admin password for the GUI.

(5) After the reboot, configure your correct timezone: /root/timezone-setup

Be advised that, when you log into the Issabel web interface, you will be prompted (three times) for your admin credentials. You can save these entries to avoid having to repeat it in the future. Now you can jump over to the Incredible PBX for Issabel tutorial to complete your installation. Within a couple minutes, your PBX will be ready to accept calls. Enjoy!

Published: Monday, August 7, 2017  


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 PBX in a Flash 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.
 



Some Recent Nerd Vittles Articles of Interest…

Almost Free: Professional Grade TTS Comes to Issabel 4



There’s no need to be chained to your TV for breaking news and weather forecasts when they can be as close as the nearest VoIP phone. Today we’re elevating text to speech with Issabel to commercial-quality. We’re wrapping up our month-long romance with Issabel 4 by introducing IBM’s Bluemix TTS service for Incredible PBX®. It’s surprisingly affordable. The first million characters of text-to-speech synthesis are FREE every month so, for most users, upgrading to commercial quality speech synthesis is a no-brainer. Try out our 10-second demo and prepare to be amazed. We provided a plain text demo (without any voice transformation SSML) to show how incredibly accurate IBM’s basic voice synthesis engine is. With additional tweaking using IBM’s SSML functions, any voice nuances can be quickly corrected or enhanced. Feel free to build a few samples on your own at IBM’s demo site.


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An awesome text-to-speech engine, of course, is only half of the story. You still need application software to bring TTS to life on your PBX. Nerd Vittles tried and true news and weather applications for Incredible PBX provide the glue that binds news and weather updates to your phone by simply dialing a 3-digit extension on your PBX. 951 gets you the latest breaking news from Yahoo, and 947 gets you current weather conditions and a weather forecast for any zip code in the United States. It’s pure, open source GPL code so feel free to modify it to meet your needs. Additional weather data is available from IBM Bluemix at modest cost for our international friends. Make that your weekend project!

Getting Started with IBM Bluemix TTS Service

NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:

You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with Incredible PBX for Issabel. Start by logging in to your IBM Bluemix account. Once you’re logged in, click on your account name (1) in the upper right corner of your web page to reveal the pull-down to select your Region, Organization, and Space. Follow the blue links at the bottom of the pull-down menu to create an Organization and Space for your TTS service.



Next, click the Menu icon which is displayed as three horizontal bars on the left side of the web page. Choose Watson. Click Create Watson Service and select Text to Speech from the applications listing. Watson will generate a new TTS service template and display it. Make certain that your Region, Organization, and Space are shown correctly. Then verify that the Standard Pricing Plan is selected. When everything is correct, click the Create button.

When your Text to Speech application displays, click Service Credentials and then click New Credential (+). When the Add New Credential dialog appears, leave the default settings as they are and click Add. Your Credentials Listing then will appear. Click View Credentials beside the new entry you just created. Write down your URL, username, and password. You’ll need these to configure the IBM Bluemix TTS service in Issabel momentarily. Logout of the IBM Cloud by clicking on the little face in the upper right corner of your browser window and choose Log Out. Confirm that you do, indeed, wish to log out. NOTE: For new implementations, you will have an APIkey instead of a username and password.

Implementing IBM Bluemix TTS Service with Issabel

Now for the fun part. We’ve built all the pieces you’ll need to deploy IBM’s TTS service and to reconfigure the Incredible PBX news and weather applications to take advantage of IBM’s new text synthesis engine. There are 5 Simple Steps to put all the pieces in place for this. Begin by (1) installing Issabel 4 on your favorite platform. Next, (2) install Incredible PBX for Issabel by following our tutorial. Now (3) log into your Issabel PBX as root using SSH or Putty and issue the following commands:

cd /var/lib/asterisk/agi-bin
wget http://incrediblepbx.com/ibmtts-issabel.tar.gz
tar zxvf ibmtts-issabel.tar.gz
nano -w /var/lib/asterisk/agi-bin/ibmtts.php

When the installation finishes, (4) an editor will open to let you insert your IBM Bluemix TTS credentials. Do so and then press Ctrl-X, Y, and Enter to save your entries. For new deployments, your API Username will be apikey, and your API Password will be your actual APIkey. Finally, while still in the agi-bin directory, (5) run the following script to update your Asterisk dialplan: ./install-ibmtts-dialplan.sh.

Now you’re ready to take IBM’s Bluemix TTS for a test drive. Pick up any phone connected to your PBX and dial 951 for the latest Yahoo news. Then dial 947 and enter a 5-digit zip code to retrieve the latest weather conditions and weather forecast for your zip code. Enjoy!

If you’d like to try out the News application with IBM Bluemix, feel free call our Demo PBX and choose option 5:

Published: Monday, July 31, 2017  


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 PBX in a Flash Forum.


 

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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.
 



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