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Free Calling Returns with Incredible PBX 15 for Wazo
Since we began our XiVO/Wazo adventure 18 months ago, the most requested feature has been direct support for Google Voice with OAuth 2 authentication. For those in the United States, it remains the cheapest VoIP solution on the planet with unlimited free calls in the U.S. and Canada. When Wazo migrated to Asterisk® 15, the Asterisk platform natively supported Google Voice, but there was no direct support in either Wazo or Incredible PBX. Instead, over the past couple months, we’ve had Google Voice functionality through Simonics SIP to Google Voice Gateway. Well, today’s a new day. We’re pleased to announce Google Voice is back!
Overview. If you’re new to Google Voice, here’s how the installation scenario goes. First, you set up a Gmail account at gmail.com. Next, you create a Google Voice account. Then, you configure Google Voice for use with Asterisk®. Next, you obtain your Google Voice OAuth 2 Refresh Token which becomes your password to use in configuring Google Voice on the Wazo platform. Next, using SSH or Putty, you log into your Wazo server as root and run the add-gvtrunk script to get your Google Voice credentials set up in Wazo. Finally, you log into the Wazo GUI with a browser and set up a custom trunk as well as an outgoing and incoming route for Google Voice calls. To add more Google Voice trunks, you simply repeat the drill. You now should have a perfectly functioning, free VoIP platform compliments of Google and Sylvain Boily and his development team. Start by deploying Incredible PBX 15 for Wazo.
Configuring Google Voice for Wazo
If you’re one of the five people on Earth that does not yet have a Gmail account, start there. Once you’ve set up your Gmail account and logged in, open a new browser tab to access the Google Voice site. Accept the Google Terms and Privacy Policy. Then choose a new Phone Number in your favorite area code. NOTE: Before Google will assign you a number, you must enter an existing U.S. phone number to verify your identity and location as well as to use for initially forwarding calls. Once your account is set up, you will get an email asking that you verify your email address.
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 http://google.com/voice. 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.
Next, click on the Calls tab. Make sure your settings match these:
- Call Screening – OFF
- Call Presentation – OFF
- Caller ID (In) – Display Caller’s Number
- Caller ID (Out) – Don’t Change Anything
- Do Not Disturb – OFF
- Call Options (Enable Recording) – OFF
- Global Spam Filtering – ON
Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Then click Save Settings. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.
One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in your Google Voice password, and don’t enable two-step authentication.
Now it’s time to obtain your OAuth 2 credentials. Even though it’s a bit more work on the front end, the good news is you won’t have to worry about your Google Voice trunks failing when Google phases out plain-text passwords. The other good news is you won’t be passing your plain-text Google Voice credentials across the Internet for everyone in the world to see.
Obtaining Your Google Voice OAuth 2 Credentials
While you’re still logged into your Google Voice account, you need to obtain a refresh_token which is what you’ll use instead of a password when setting up your Google Voice account with XiVO. Here’s how.
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.
With a recent Google change, you then will be prompted to choose your Google account again. Click on the same account name that you logged into in Step #1. You then will be prompted whether to allow Google to view and send Chat messages. Click Allow.
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 with the add-gvtrunk script. 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.
Downloading and Installing Google Voice with OAuth 2
Installs of Incredible PBX 15 for Wazo beginning December 7 already have the necessary components in place to support Google Voice. If you have an earlier installation of Incredible PBX 15 for Wazo, you can easily upgrade your server to support Google Voice. Simply log into your server’s Linux CLI as root using SSH or Putty and issue the following commands:
cd /root wget http://incrediblepbx.com/gvtrunk15.tar.gz tar zxvf gvtrunk15.tar.gz rm -f gvtrunk15.tar.gz
If you’re upgrading Wazo from a release before 17.15, you can follow the usual upgrade steps after first removing your existing Google Voice trunks from the configuration. Start by copying your Google Voice credentials from /etc/asterisk/oauth_xmpp.conf. Next, delete the following file: /etc/asterisk/extensions_extra.d/gv.conf. Then perform the upgrade:
cd /root cp -pr /etc/postfix /root cp -p /etc/nginx/sites-enabled/xivo /root echo "noload => res_xmpp_oauth.so" >> /etc/asterisk/modules.conf xivo-dist wazo-17.17 apt-get update /etc/init.d/netfilter-persistent stop wazo-upgrade iptables-restart # put the Incredible PBX web add-ons back in place cd / wget http://incrediblepbx.com/incredible-nginx.tar.gz tar zxvf incredible-nginx.tar.gz rm -f incredible-nginx.tar.gz ln -s /etc/nginx/locations/https-available/01_incrediblepbx /etc/nginx/locations/https-enabled/. cd /etc/nginx wget http://incrediblepbx.com/nginx-config.tar.gz tar zxvf nginx-config.tar.gz rm -f /etc/nginx/sites-enabled/default cp -p /root/xivo /etc/nginx/sites-enabled /etc/init.d/nginx restart cp -pr /root/postfix /etc /etc/init.d/postfix restart sed -i 's|fr|us|' /etc/asterisk/indications.d/99-general.conf /etc/init.d/asterisk restart
Initial Google Voice OAuth Trunk Setup with Wazo
Now it’s time to create your Google Voice OAuth 2 trunk to use with Wazo. Log back into the Linux CLI as root. Be sure you have your 10-digit Google Voice phone number, your Google account name without @gmail.com, and your Refresh Token from the previous step. Then run the script to add your Google Voice trunk:
cd /root ./add-gvtrunk
Plug in your Google Voice phone number, your account name (without @gmail.com), and refresh token when prompted. Check your work carefully. When the install finishes, fire up your favorite browser to finish the setup using the settings displayed in the Linux CLI.
Configuring Wazo for Google Voice OAuth
From a browser pointed to your server’s IP address, log in to Wazo as root with your GUI password.
Choose Services.IPBX.Trunk Management.Customized. Click on + Add to create a new custom trunk. Configure the trunk using the settings provided in the previous step. Click the Save button when finished. The Trunk Name will be your actual gmailname (without @gmail.com). Interface will be Motif/gmailname (using your actual Gmail name). Interface suffix will be @voice.google.com. And the Context will be Outcalls (to-extern).
Next, choose Services.IPBX.Call Management.Incoming Calls. Click on + Add to create a new inbound route for your Google Voice DID. This is where you tell Wazo how to route calls placed to your Google Voice number. For your DID, enter your 10-digit Google Voice number. For the Context, choose Incalls (from-extern). Then choose from the pick lists to select a Destination and Redirect option for the incoming calls. It could be an extension, a ring group, a conference room, or an IVR. Click Save when you’re finished.
Finally, choose Services.IPBX.Call Management.Outgoing Calls. Click on + Add to create a new outbound route. Under the General tab, configure the route as shown below using a Name of out_gmailname (using your actual Gmail name). The Context should be Outcalls (to-extern). The Preprocess subroutine should be subr-gv-outcall. And the desired Custom Trunk should be dragged left to the selected column:
Under the Exten tab, add the desired 10-digit Exten string that users will dial to place a call through the Google Voice trunk. Trim off any prefix (such as 9) that you assigned to the dial string using the Stripnum setting. Then click the Save button.
For example, for a first trunk, you might choose NXXNXXXXXX as the Exten with a Stripnum of 0. This would tell Wazo to route all 10-digit calls to this Custom GV Trunk. For a second Google Voice trunk, you might choose 9NXXNXXXXXX with a Stripnum of 1. This would tell Wazo to route 11-digit calls with a 9 prefix to this Custom Trunk AND to strip off the first digit (9) before sending the 10-digit call to Google Voice.
SMS Messaging with Google Voice
Incredible PBX implementations include a separate Python-based component called pygooglevoice that allows SMS messaging using a Google Voice account. You’ll find the SMS Dictator and SMS message blasting scripts in the /root folder of your server. Be advised that the gvoice script used to send these SMS messages does NOT use OAuth 2 authentication and requires your traditional Google Voice account name and password. Because it uses plain text, you probably will experience failures using gvoice unless your PBX is housed on the same server from which you traditionally access this account. The workaround is to do the following immediately before attempting to use SMS messaging with Google Voice. Using a browser, log into the Gmail account that corresponds to the account you plan to use for SMS messaging. In a separate tab of your browser while still logged into Gmail, go to the following link and enable Less Secure Apps for this account. In a third tab, go to the following link and activate the Google Reset Procedure. Now, from the Linux CLI on your PBX, attempt to login to gvoice using your Gmail account name including @gmail.com AND your plain-text Gmail password. If it succeeds, you’re good to go and can now use the SMS messaging tools. Type quit to exit from gvoice. Google normally follows up insecure access attempts with an email requesting that you confirm this is a "safe IP address" for your account. If you fail to respond to that email, you probably will need to repeat this exercise every time you attempt to use SMS messaging.
Taking Google Voice for a Test Drive
That completes the Google Voice setup. You now should be able to place a call using your Google Voice trunk by dialing any 10-digit number. And calls placed to your Google Voice number should ring at the inbound destination you chose above.
If you have additional Google Voice trunks, simply run /root/add-gvtrunk again and insert the new credentials.
If you ever need to delete a Google Voice account, just run /root/del-gvtrunk with the name of the Google Voice trunk to delete. Enjoy your free phone service!
Originally published: Monday, December 11, 2016
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 and Wazo 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 or Wazo? Visit the PBX in a Flash Forum.
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Some Recent Nerd Vittles Articles of Interest…
It’s Soup: The Definitive Quick Start Guide to Wazo 17.17
Today we’re pleased to introduce Wazo 17.17, the latest iteration in the Wazo fork of XiVO. Twelve years in the making with the same development team, Wazo 17.17 is really something special. It’s the latest release featuring Asterisk® 15 with multi-party videoconferencing and also includes a new easy-to-use WebRTC feature and support for Asterisk FollowMe Roaming. We are pleased to announce that this release of Wazo 17.17 using the latest Incredible PBX 15 installer for Asterisk 15 now supports native Google Voice trunks with OAuth 2. Of course, all of your favorite Incredible PBX add-ons are ready as well. What follows is a new soup-to-nuts tutorial covering everything you need to know to get started with Incredible PBX for Wazo 17.17.
Wazo 17.17 enhances the brand new user interface to complement Wazo’s existing GUI. Revolutionary is probably the best way to describe its design. The reason is that it is generated using pure API calls to the Wazo engine. What that means to the administrator is you can tweak it, enhance it, replace it, or build additional functionality into your UI using the same API calls that the Wazo developers employed to build these components. This is obviously the first of many iterations that will be released in coming months, but it gives you an opportunity to kick the tires. Once you complete the installation process which follows, simply use a browser to log into the IP address of your server at the following link: https://serverIP/admin/. To begin, login with your root GUI credentials and click Plugins to display available options (pictured above). Then install each of the following plugins to begin: Users, Extensions, Contexts, and Devices. The remaining plugins are optional; however, the following plugins are not yet operational: Outcalls and Trunks. The Parking Lots and Switchboard plugins are functional but do not yet show existing setups configured through the legacy GUI. Additional tips and getting started tips are available here and here.
If you’ve been following along in our XiVO adventure with Incredible PBX last year, then you already know that there were a significant number of configuration hoops to jump through once the base install was finished. While these steps are well documented in the original Incredible PBX for XiVO tutorial, there still were plenty of opportunities for typos and skipping steps. Any misstep could spell the difference in a perfectly functioning PBX and one that couldn’t make or receive calls. So we introduced XiVO Snapshots, and today we bring the same technology to the Wazo platform as well.
A Wazo Snapshot is just what the name implies. It’s a snapshot of a working Wazo PBX that has virtually everything already configured: SIP settings that work with Asterisk®, a SIP extension that works with a SIP phone plus your cellphone, a SIP extension preconfigured for WebRTC that uses the new Opus codec, SIP and Google Voice trunk setups for many of the major commercial providers, and default inbound and outbound routes to ease the task of routing calls into and out of your PBX. Basically, you plug in your credentials from your favorite provider after running the Incredible PBX for Wazo installer, tell Wazo how to route the calls, and you’re done. You can have a stable and functional PBX making calls to anywhere in the world in a matter of minutes. Then you can review our numerous tutorials to add additional bells and whistles while actually using your Wazo PBX while you learn.
Incredible PBX for Wazo Installation Overview
Before we roll up our sleeves and walk you through the installation process, we wanted to provide a quick summary of the 10 Basic Steps in setting up Incredible PBX for Wazo. By the way, the whole process takes less than an hour!
- Set Up Desired PBX Platform: Stand-alone PC, Virtual Machine, or Cloud-Based Server
- Run the Incredible PBX for Wazo installer
- Set Up One or More SIP or Google Voice Trunks for Your PBX
- Tell Wazo Where to Direct Incoming Calls from Each Trunk
- Tell Wazo Which Trunk to Use for Every Outbound Calling Digit Sequence
- Set Up a SoftPhone or WebRTC Phone (or both)
- Decide Whether to Activate Simultaneous Ringing on your Cellphone
- Add Google Speech Recognition Key (if desired)
- Activate DISA with Incredible PBX for Wazo (if desired)
- Test Drive Incredible PBX for Wazo
1. Incredible PBX for Wazo Hardware Platform Setup
NOTE: Ignore references to Debian 9 in the platform setup tutorials. Debian 9 is not quite ready yet so stick with Debian 8.
The first step is to choose your hardware platform and decide whether you want to babysit a server and network or leave those tasks to others. We’ve taken the guesswork out of the setups documented below. Five of the options are cloud providers, each of whom provides a generous discount to let you kick the tires. So click on the links below to review the terms and our walkthrough of the setup process on each platform.
- Wazo Platform Tutorial: Installing Wazo on a Dedicated PC
- Wazo Platform Tutorial: Installing Wazo VM on VirtualBox
- Wazo Platform Tutorial: Installing Wazo VM on VMware ESXi
- Wazo Platform Tutorial: Installing Wazo VM at ImpactVPS
- Wazo Platform Tutorial: Installing Wazo VM at Vultr
- Wazo Platform Tutorial: Installing Wazo VM at OVH.com
- Wazo Platform Tutorial: Installing Wazo VM at Digital Ocean
- Wazo Platform Tutorial: Installing Wazo VM at WootHosting
- Wazo Platform Tutorial: Installing Wazo VM at CloudAtCost
- Wazo Platform Tutorial: Manually Installing Wazo and Debian 8
2. Running the Incredible PBX for Wazo Installer
Once you have your hardware platform up and running, the rest of the initial setup process is easy. Simply download and run the Incredible PBX for Wazo installer. On some platforms, it first updates Debian 8 to current specs and reboots. Then log back in and rerun the installer a second time if you are prompted to do so.
cd /root wget http://incrediblepbx.com/IncrediblePBX15-Wazo.sh chmod +x IncrediblePBX15-Wazo.sh ./IncrediblePBX15-Wazo.sh
3. Setting Up SIP and Google Voice Trunks with Wazo
When the installation is finished, you can make toll-free calls in the U.S. and Canada without doing anything except dialing "1″ and the 10-digit number from any phone connected to your server. For other calls, there are two steps in setting up trunks to use with Incredible PBX. First, you have to sign up with the provider of your choice and obtain trunk credentials. These typically include the FQDN of the provider’s server as well as your username and password to use for access to that server. Second, you have to configure a trunk on the Incredible PBX for Wazo server so that you can make or receive calls outside of your PBX. As with the platform tutorials, we have taken the guesswork out of the trunk setup procedure for roughly a dozen respected providers around the globe. In addition, Wazo Snapshots goes a step further and actually creates the trunks for you, minus your credentials, as part of the initial Incredible PBX install.
For Google Voice trunks with Incredible PBX 15, log into your server as root and run ./add-gvtrunk. When prompted, insert your 10-digit Google Voice number, your Google Voice email address and your OAuth 2 token. The native Google Voice OAuth tutorial explains how to obtain it.
For the other providers, review the setup procedure below and then edit the preconfigured trunk for that provider by logging into the Wazo web GUI and choosing IPX → Trunk Management → SIP Protocol. Edit the setup for your provider (as shown above) and fill in your credentials and CallerID number in the General tab. Activate the trunk in the Register tab after again filling in your credentials. Save your settings when finished. No additional configuration for these providers is required using the Incredible PBX for Wazo Snapshot.
- Wazo Trunks Tutorial: Installing a Vitelity SIP Trunk
- Wazo Trunks Tutorial: Installing a VoIP.ms SIP Trunk
- Wazo Trunks Tutorial: Installing a RingPlus SIP Trunk
- Wazo Trunks Tutorial: Installing a FreeVoipDeal (Betamax) SIP Trunk
- Wazo Trunks Tutorial: Installing a Google Voice-Simonics SIP Trunk
- Wazo Trunks Tutorial: Deploying Native Google Voice with OAuth Trunks
- Wazo Trunks Tutorial: Installing an Anveo Direct Outbound SIP Trunk
- Wazo Trunks Tutorial: Installing a Skype Connect SIP Trunk
- Wazo Trunks Tutorial: Installing a LocalPhone SIP Trunk
- Wazo Trunks Tutorial: Installing a CallCentric SIP Trunk
- Wazo Trunks Tutorial: Installing a FlowRoute SIP Trunk
4. Directing Incoming Calls from Wazo Trunks
Registered Wazo trunks typically include a DID number. With the exception of CallCentric, this is the number that callers would dial to reach your PBX. With CallCentric, it’s the 11-digit account number of your account, e.g. 17771234567. In the Wazo web GUI, we use IPX → Call Management → Incoming Calls to create inbound routes for every DID and trunk associated with your PBX. Two sample DIDs have been preconfigured to show you how to route calls to an extension or to an IVR. To use these, simply edit their settings and change the DID to match your trunk. Or you can create new incoming routes to send calls to dozens of other destinations on your PBX.
5. Routing Outgoing Calls from Wazo to Providers
Outgoing calls from extensions on your Wazo PBX must be routed to a trunk provider to reach call destinations outside your PBX. Outgoing call routing is managed in IPX → Call Management → Outgoing Calls. You tell Wazo which trunk provider to use in the General tab. Then you assign a Calling Digit Sequence to this provider in the Exten tab. For example, if NXXNXXXXXX were assigned to Vitelity, this would tell Wazo to send calls to Vitelity if the caller dialed a 10-digit number. Wazo has the flexibility to add and remove digits from a dialed number as part of the outbound call routing process. For example, you might want callers to dial 48NXXNXXXXXX to send calls to a Google Voice trunk where 48 spells "GV" on the phone keypad. We obviously don’t want to send the entire dial string to Google Voice so we tell Wazo to strip the first 2 digits (48) from the number before routing the call out your Google Voice trunk. We’ve included two examples in the Wazo Snapshot to get you started. Skype Connect (shown below) is an example showing how to strip digits and also add digits before sending a call on its way:
6. Setting Up a Softphone & WebRTC with Wazo
If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. In the World of Wazo, you’ll find these under IPBX → Services → Lines. Just click on the Pencil icon beside the extension to which you want to connect. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (4871) to try things out. With Telephone, you can use over two dozen soft phones simultaneously on your desktop.
For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Wazo Line. You’ll need the IP address of your server plus your Line username and password associated with the 701 extension. On the Wazo platform, do NOT use an actual extension number for your username with Wazo. Go to IPBX Settings → Lines to decipher the appropriate username and password for the desired extension. Click OK to save your entries.
WebRTC allows you to use your Chrome or Firefox browser as a softphone. Extension 701 comes preconfigured for WebRTC access with Incredible PBX for Wazo. It shares the same password as the Line associated with extension 701, but the username is 701 rather than the username associated with the Line. You can decipher the password by accessing the Wazo Web GUI and then IPBX → Services → Users → Incredible PBX → XiVO Client Password. Or log into your server as root using SSH or Putty and run: /root/show-701-pw
. Wazo introduces several new features to WebRTC including support for the awesome new Opus codec plus voicemail management and even Gravatar support. It’s all preconfigured!
Special Note: Beginning with version 17.02 of Wazo, WebRTC is fully integrated with NGINX on your server, and a simplified method for configuring WebRTC users has been added. When you create a new User account, simply choose the SIP (WebRTC) Protocol when creating a new user account, and all of the Advanced Line options required to support WebRTC will be preconfigured for you.
To use WebRTC, you no longer need to accept the different SSL certificates associated with the WebRTC app. From your browser, go to the following site and click on each link to accept the certificates. Once you’ve completed this process, simply visit the Wazo WebRTC site.
Before logging in, click on the Gear icon in the lower right corner and then click on the Pencil icon to edit your Settings. Fill in the public IP address of your Wazo server and specify 443 for the Port. Leave the Backend field blank and click Save. Now login to your WebRTC account with Username 701. The Password is the one you obtained running show-701-pw. The IP Address (if required) is the address of your Wazo PBX.
Implementing FollowMe Roaming with a CellPhone
In addition to ringing your SIP extension when incoming calls arrive, Wazo 17.17 can also ring your cellphone simultaneously. This obviously requires at least one outbound trunk. If that trunk provider also supports CallerID spoofing, then Wazo will pass the CallerID number of the caller rather than the DID associated with the trunk. Incredible PBX for Wazo comes with cellphone support for extension 702 ready to go. To enable it, access the Wazo Web GUI and go to IPBX → Services → Users → Incredible PBX and insert your Mobile Phone Number using the same dial string format associated with the trunk you wish to use to place the calls to your cellphone. You then can answer the incoming calls on either your cellphone or the registered SIP phone. If you answer on your cellphone, you will be prompted whether you wish to accept the call. If you press 1 after observing the CallerID, the caller will be connected. If you decline, the caller will be routed to the Wazo voicemail account of the extension.
8. Activating Voice Recognition for Wazo
Google has changed the licensing of their speech recognition engine about as many times as you change diapers on a newborn baby. Today’s rule restricts use to “personal and development use.” Assuming you qualify, the very first order of business is to enable speech recognition for your Wazo PBX. Once enabled, the Incredible PBX feature set grows exponentially. You’ll ultimately have access to the Voice Dialer for AsteriDex, Worldwide Weather Reports where you can say the name of a city and state or province to get a weather forecast for almost anywhere, Wolfram Alpha for a Siri-like encyclopedia for your PBX, and Lefteris Zafiris’ speech recognition software to build additional Asterisk apps limited only by your imagination. And, rumor has it, Google is about to announce new licensing terms, but we’re not there yet. To try out the Voice Dialer in today’s demo IVR, you’ll need to obtain a license key from Google. This Nerd Vittles tutorial will walk you through that process. Add your key to /var/lib/asterisk/agi-bin/speech-recog.agi on line 72.
9. Adding DISA Support to Your Wazo PBX
If you’re new to PBX lingo, DISA stands for Direct Inward System Access. As the name implies, it lets you make calls from outside your PBX using the call resources inside your PBX. This gives anybody with your DISA credentials the ability to make calls through your PBX on your nickel. It probably ranks up there as the most abused and one of the most loved features of the modern PBX.
There are three ways to implement DISA with Incredible PBX for Wazo. You can continue reading this section for our custom implementation with two-step authentication. There also are two native Wazo methods for implementing DISA using a PIN for security. First, you can dedicate a DID to incoming DISA calls. Or you can add a DISA option to an existing IVR. Both methods are documented in our tutorial on the PIAF Forum.
We prefer two-step authentication with DISA to make it harder for the bad guys. First, the outside phone number has to match the whitelist of numbers authorized to use your DISA service. And, second, you have to supply the DISA password for your server before you get dialtone to place an outbound call. Ultimately, of course, the monkey is on your back to create a very secure DISA password and to change it regularly. If all this sounds too scary, don’t install DISA on your PBX.
1. To get started, edit /root/disa-xivo.txt. When the editor opens the dialplan code, move the cursor down to the following line:
exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1) ; Good guy
2. Clone the line by pressing Ctrl-K and then Ctrl-U. Add copies of the line by pressing Ctrl-U again for each phone number you’d like to whitelist so that the caller can access DISA on your server. Now edit each line and replace 701 with the 10-digit number to be whitelisted.
3. Move the cursor down to the following line and replace 12341234 with the 8-digit numeric password that callers will have to enter to access DISA on your server:
exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)
4. Save the dialplan changes by pressing Ctrl-X, then Y, then ENTER.
5. Now copy the dialplan code into your Wazo setup, remove any previous copies of the code, and restart Asterisk:
cd /root sed -i '\:// BEGIN DISA:,\:// END DISA\:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf cat disa-xivo.txt >> /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf /etc/init.d/asterisk reload
6. The traditional way to access DISA is to add it as an undisclosed option in an IVR that is assigned to one of your inbound trunks (DIDs). For the demo IVR that is installed, edit the ivr-1.conf configuration file and change the "option 0″ line so that it looks like this. Then SAVE your changes.
exten => 0,1(ivrsel-0),Dial(Local/3472@default)
7. Adjust the inbound calls route of one of your DIDs to point to the demo IVR by changing the destination to Customized with the following Command:
Goto(ivr-1,s,1)
A sample is included in the Wazo Snapshot. Here’s how ours looks for the Demo IVR:
8. Now you should be able to call your DID and choose option 0 to access DISA assuming you have whitelisted the number from which you are calling. When prompted, enter the DISA password you assigned and press #. You then should be able to dial a 10-digit number to make an outside call from within your PBX.
SECURITY HINT: Whenever you implement a new IVR on your PBX, it’s always a good idea to call in from an outside number 13 TIMES and try every key from your phone to make sure there is no unanticipated hole in your setup. Be sure to also let the IVR timeout to see what result you get.
10. NEW: Travelin’ Man 3 WhiteList Firewall
All new installs of Incredible PBX for Wazo include the newly enhanced and preconfigured IPtables Firewall featuring Travelin’ Man 3. Please read the new tutorial when you have a few minutes. It documents how to enable access to your server for remote phones and traveling users.
PortKnocker previously let a remote user ping the server with a 3-digit code to temporarily add the remote user’s IP address to the IPtables Firewall WhiteList so that the user could gain remote access. We now are adding the ability to let end-users make successful PortKnocks permanent just as if an administrator had used add-ip on the server to add an IP address. Administrators still can remove these entries using del-acct although you will have to know the date and time that the entry was added. The file names consist of timestamp.iptables in /root and the additions are made to the custom WhiteList in /usr/local/sbin/iptables-custom. To enable permanent PortKnocker additions, simply run this command while logged into your server as root: iptables-knock activate.
11. Test Drive Incredible PBX for Wazo
To give you a good idea of what to expect with Incredible PBX for Wazo, just pick up a phone and dial any toll-free number in the U.S. and Canada using a 1 prefix. We’ve also set up a sample IVR using voice prompts from Allison. Try it out from any phone on your PBX by dialing 4871 (IVR1):
Nerd Vittles Demo IVR Options
1 – Call by Name (say "Delta Airlines" or "American Airlines" to try it out)
2 – MeetMe Conference
3 – Wolfram Alpha (Coming Soon!)
4 – Lenny (The Telemarketer’s Worst Nightmare)
5 – Today’s News Headlines
6 – Weather Forecast (enter a 5-digit ZIP code)
7 – Today in History (Coming Soon!)
8 – Speak to a Real Person (or maybe just Lenny if we’re out)
What To Do and Where to Go Next?
Here are a boatload of projects to get you started exploring Wazo on your own. Just plug the keywords into the search bar at the top of Nerd Vittles to find numerous tutorials covering the topics or simply follow our links. Unless there is an asterisk (*) the components already are in place so do NOT reinstall them. Just read the previous tutorials to learn how to configure each component. Be sure to also join the PIAF Forum to keep track of the latest tips and tricks with Wazo. There’s a treasure trove of information that awaits.
- Deploy Gmail as SMTP Relay for Wazo
- Build IVRs in Minutes
- Activate SMS Messaging
- Configure CallerID Superfecta
- Explore Munin Graphics in Wazo
- Integrate Google Calendar Alerts
- Configure NeoRouter and PPTP VPN
- Set Up EndPoint Management in Wazo
- Customize and Create New Wazo IVRs
- Try Out SQLite3 Call Detail Reports
- Learn About Wazo Backups and Upgrades
- Deploy SIP URIs for Free Worldwide Calling
- Develop a PortKnocker Emergency Access Plan
- Deploy Redundancy with Wazo High Availability
- * Install FCC RoboCall BlackList with WhiteList
- * Install Siri-Like Wolfram Alpha Module
- * Install Lenny, The Robocallers’ Worst Nightmare
Wazo and Incredible PBX Dial Code Cheat Sheets
Complete Wazo documentation is available here. But here are two cheat sheets in PDF format for Wazo Star Codes and Incredible PBX Dial Codes.
Published: Wednesday, December 14, 2016 Last Updated: Monday, December 12, 2017
Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number.
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.
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…
The Ultimate Voice Dialer for Asterisk and Incredible PBX
Let’s face it. Voice recognition with Google has been hit and miss, and that’s on a good day. So we’re delighted to shift gears and introduce a new platform powered by IBM Watson’s Speech-to-Text (STT) engine. While it’s not free, that’s really theoretical for most of our readers. Your first month on the platform is entirely free. And, after that, you get 1,000 minutes a month of free voice recognition services. If you still want more, it’s 2¢ a minute.
We first introduced IBM’s STT platform back in March when we documented how to use the service to transcribe voicemails and deliver them via email. Today, we’re introducing the Incredible Voice Dialer for Asterisk. It runs on all of the major Incredible PBX platforms: CentOS, Wazo, and Issabel. It’s married to our AsteriDex phonebook application that is deployed with Incredible PBX using MySQL, MariaDB, or SQLite3 depending upon platform.
The way it works is a user picks up an extension on your PBX and dials 411. The caller will be prompted for the name of the person or company to call. Once the caller says the name, the Incredible Voice Dialer will send the recording to IBM’s Watson STT engine for transcription. The result is then passed to AsteriDex where the text will be matched against the phone number saved for that person or company. The number is then passed to your default outbound trunk to place the call. All of the magic happens in less than two seconds, and the call begins ringing at your destination. You can try it out for yourself on our demo server this week. Just dial: , choose option 1 when the IVR answers, and then say "Delta Airlines" or "American Airlines" when prompted for a name. The queries support wildcard matching. If you say "Delta", you’ll still be connected to Delta Airlines.
What About the Quality? Here’s the bottom line. Speech recognition isn’t all that useful if it fails miserably in recognizing everyday speech. The good news is that IBM Watson’s speech recognition engine is now the best in the business. If you want more details, read the article below which will walk you through IBM’s latest speech recognition breakthrough:
Why IBM's speech recognition breakthrough matters for AI and IoT. Via @techrepublic https://t.co/AJi8MA3E20
— IBM Developer (@IBMDeveloper) March 15, 2017
Creating an IBM Bluemix Speech to Text Account
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
1. Create Bluemix account here.
2. Confirm your registration by replying to email from IBM.
3. Login to Bluemix using your new credentials.
4. Agree to terms and conditions, name your organization, and name your space (STT).
5. Choose Watson Speech to Text service and click Create.
6. When Speech to Text-kb opens, click Service Credentials tab (on the left).
7. In Actions column, click View Credentials. Write down your username and password.
8. Logout by clicking on image icon in upper right corner of dialog window.
Install Voice Dialer with Incredible PBX for Wazo
1. Login to your server as root using SSH/Putty and issue the following commands:
cd / wget http://incrediblepbx.com/ibmstt-411-wazo.tar.gz tar zxvf ibmstt-411-wazo.tar.gz rm -f ibmstt-411-wazo.tar.gz sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf sed -i '/\\[xivo-extrafeatures\\]/r /tmp/411.txt' /etc/asterisk/extensions_extra.d/xivo-extrafeatures.conf asterisk -rx "dialplan reload"
2. Edit /var/lib/asterisk/agi-bin/getnumber.sh and insert your IBM credentials from step #7 above into these variables:
API_USERNAME="XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX" API_PASSWORD="XXXXXXXXXXXX"
3. Save the file.
Install Voice Dialer on Other Incredible PBX Platforms
1. Login to your server as root using SSH/Putty and issue the following commands:
cd / wget http://incrediblepbx.com/ibmstt-411.tar.gz tar zxvf ibmstt-411.tar.gz rm -f ibmstt-411.tar.gz sed -i '\\:// BEGIN Call by Name:,\\:// END Call by Name:d' /etc/asterisk/extensions_custom.conf sed -i '/\\[from-internal-custom\\]/r /tmp/411.txt' /etc/asterisk/extensions_custom.conf asterisk -rx "dialplan reload"
2. Edit /var/lib/asterisk/agi-bin/getnumber.sh and insert your IBM credentials from step #7 above into these variables:
API_USERNAME="XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX" API_PASSWORD="XXXXXXXXXXXX"
3. Save the file.
Take Incredible Voice Dialer for a Test Drive
1. From an extension connected to your PBX, dial 411. When prompted for the name to call, say "Delta Airlines" or "American Airlines."
2. Quicker than you could actually dial the number, you’ll be connected.
Building Voice-Enabled Applications with Asterisk
All of our code is open source, GPL2 code so you’re more than welcome to use it, learn from it, and then build your own voice-enabled applications. Just abide by the terms of the license and share. When you review /var/lib/asterisk/agi-bin/getnumber.sh, you’ll see that it’s incredibly easy to change the backend database. Here’s the Wazo flavor of the script:
API_USERNAME="XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX" API_PASSWORD="XXXXXXXXXXXX" thisfile="$1" # sending the recording to IBM Watson for transcription curl -k -u $API_USERNAME:$API_PASSWORD -X POST --limit-rate 40000 --header "Content-Type: audio/wav" --data-binary @/tmp/$thisfile.wav "https://stream.watsonplatform.net/speech-to-text/api/v1/recognize?continuous=true&model=en-US_NarrowbandModel" 1>/tmp/$thisfile.txt # grabbing the text out of the IBM Watson response msg=`cat /tmp/$thisfile.txt | grep transcript | cut -f 2 -d ":" | cut -f 2 -d '"' | sed 's| *$||' | sed -e "s/\b\(.\)/\u/g"`% # passing text to MySQL (1st line) or SQLite3 (2nd line) for name lookup. answer is num2call. #num2call=$(mysql -uroot -ppassw0rd asteridex -ss -N -e "SELECT user1.out FROM user1 where name LIKE '$msg'"); num2call=`/usr/bin/sqlite3 /var/lib/asterisk/agi-bin/asteridex.sqlite "select out from user1 where name LIKE '$msg'"` # clearing out our temporary files rm -f /tmp/$thisfile.* # passing the results to the Asterisk dialplan echo "SET VARIABLE PTY2CALL "\""$msg"\""" echo "SET VARIABLE NUM2CALL "\""$num2call"\""" # we're done with the AGI bash script so let's exit gracefully exit 0
The Asterisk dialplan code could be modified for any number of applications. Here’s what it looks like on the Incredible PBX 13 platform. It’s slightly different with Wazo to accomodate their dialplan syntax.
;# // BEGIN Call by Name exten => 411,1,Answer exten => 411,n,Playback(custom/411) exten => 411,n,Set(RANDFILE=${RAND(8000,8599)}) exten => 411,n,Record(/tmp/${RANDFILE}.wav,3,10) exten => 411,n,Playback(/tmp/${RANDFILE}) exten => 411,n,AGI(getnumber.sh,${RANDFILE}) exten => 411,n,NoOp(Party to call : ${PTY2CALL}) exten => 411,n,NoOp(Number to call: ${NUM2CALL}) exten => 411,n,Goto(outbound-allroutes,${NUM2CALL},1) exten => 411,n,Hangup() ;# // END Call by Name
There’s nothing magical about it. (1) It answers the call to 411. (2) It plays back a recording that prompts the user to say the name of the person or company to call. (3) It generates a random number to use for the filenames associated with the STT process. (4) It records the caller’s speech and saves it to the random filename as a .wav file which IBM STT can understand. (5) It passes the call to the AGI bash script to send the recording to IBM Watson and obtain the transcription and to pass the text to MySQL or SQLite3 to lookup the text in the AsteriDex database. (6) We display the called party’s name on the Asterisk CLI. (7) We display the called party’s phone number on the Asterisk CLI. (8) We place the call using the PBX’s default outbound route. (9) We hangup the call when it’s completed.
Published: Monday, October 9, 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
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Some Recent Nerd Vittles Articles of Interest…
VoIP Security: Installing SSL Certificates with Incredible PBX
We’ve got some revolutionary VoIP projects coming your way over the next several weeks, but I’m sorry to say the hardest part of them is getting your server configured to use secure and encrypted web communications via HTTPS. This is quickly becoming a universal requirement of most of the major technology players. So what might not be the most glamorous VoIP topic for a Monday morning is not only necessary but long overdue. The good news is that obtaining, installing, configuring, and maintaining an SSL certificate for your VoIP server is not the royal pain that it once was. And, by this time next week, you’ll be glad you went through the exercise. Thankfully, the EFF’s Certbot project is available to assist in installing free certificates from Let’s Encrypt.
Before we begin, here’s a word to the wise. You will save yourself a thousand headaches by deploying your Incredible PBX server in the cloud where you get a dedicated IP address and can easily assign a fully-qualified domain name (FQDN) to your server. Options now are available for as little as $1.50 to $3.50/month including Vultr which provides an incredibly reliable platform in many cities for as little as $2.50 a month. And another 50¢ buys you weekly image backups without lifting a finger. They can be restored with one click! If reliability and redundancy matter, you can’t beat Vultr’s price or the feature set, and we have tutorials to get you started with either Wazo or Issabel. If cost is your sole criteria, you can’t beat WootHosting at $1.50 a month. You’ll find a tutorial here. If performance is critical, you can’t beat OVH at $3.50/month with a Wazo tutorial here and an Issabel tutorial here. Finally, if you’re technically challenged, our corporate sponsor, RentPBX, will do all of the cloud migration for you and provide a turnkey, high performance VoIP platform for just $15/month. So what are you waiting for? Now’s the time. No excuses! It’s not going to get any cheaper or more reliable. And next week you’ll be thanking us. For these reasons, we’re saying goodbye our home-based servers sitting behind NAT-based firewalls. With the projects coming down the pike, the mountain is just too steep to continue that trek unless you have the technical expertise to pull it off yourself.
Obtaining and Installing an SSL Certificate
For CentOS 6 running Incredible PBX 13 or CentOS 7 running Incredible PBX for Issabel 4, begin by making certain that you can access your site using its FQDN with HTTP, e.g. http://myserver.mydomain.org. Get that working first. Next, log into your server as root using SSH/Putty and issue the following commands:
yum -y install python-devel python-pip python-setuptools python-virtualenv --enablerepo=epel yum -y install centos-release-scl yum -y install python27 scl enable python27 bash pip -V # should show python 2.7 pip install --upgrade pip pip install requests registry urllib3 pyOpenSSL --force --upgrade pip install certbot-apache --force --upgrade cd /root wget https://dl.eff.org/certbot-auto chmod a+x certbot-auto service iptables stop ./certbot-auto --authenticator webroot --installer apache -w /var/www/html -d FQDN.here iptables-restart service httpd restart exit
During the automated setup, you can decide whether to force all web traffic to the secure site. We recommend it. Once the install finishes, test access to your server by going to your FQDN using HTTPS. Don’t continue with the setup until you get HTTPS working and your browser shows you have a SECURE site! Remember that you must renew your free certificate every 90 days by using the following /root/certbot-update script:
#!/bin/bash echo "Before you begin, type: scl enable python27 bash" echo "Then rerun this update script and press ENTER." read -p "If you already have done so, press Enter. Otherwise, Ctrl-C now" service iptables stop ./certbot-auto --authenticator webroot --installer apache -w /var/www/html -d FQDN.here iptables-restart echo "Type exit when this script completes." exit
For Debian 8 running Incredible PBX for Wazo, things are a bit more complicated because Wazo forces HTTPS access even though you do not yet have a certificate for your FQDN. Because of its NGINX web server platform, with Wazo you’ll have to manually install and configure certificates with certbot and LetsEncrypt. The silver lining with Wazo is HTTPS access gets you a WebRTC phone with a couple button clicks. Go to this link, click on the Config wheel (bottom right), click on the Pencil icon and plug in the FQDN of your server. Click SAVE. Enter your login name as 701 and the password assigned to the extension which you can obtain by running: /root/show-701-pw. That’s probably the quickest phone setup you’ll ever find. But we’re getting ahead of ourselves…
1. Let’s get certbot installed. Login to your server as root using SSH or Putty and issue the following commands:
cd /etc/apt echo "deb http://ftp.debian.org/debian jessie-backports main" >> sources.list apt-get update apt-get install certbot -t jessie-backports
2. Temporarily, turn off HTTPS since the certificate install requires HTTP access. In /etc/nginx/sites-enabled/xivo, comment out these 3 lines and save the updated file:
In server section for port 80: # include /etc/nginx/locations/http-enabled/*; In server section for port 443: # listen 443 default_server; # server_name $domain;
Then restart the web server: /etc/init.d/nginx restart. Now you have a basic http web server. If you want to verify that it’s working, use a browser and go to http://YOUR-FQDN/asteridex4/index.php. It should download the file to your desktop which isn’t desirable, but this is only temporary.
3. In /var/www/html, issue the following commands:
cd /var/www/html mkdir .well-known cd .well-known mkdir acme-challenge cd acme-challenge chown -R asterisk:www-data /var/www/html/.well-known
Leave this SSH/Putty session running temporarily and open a second SSH/Putty connection to your server logging in as root.
4. Disable your firewall temporarily: /etc/init.d/netfilter-persistent flush
5. Start the certbot installation script: certbot certonly –manual
6. You’ll be prompted for the FQDN of your server to generate the certificates. Then you’ll be given an oddball name AND an expected oddball response. With these two entries in hand, temporarily switch back to your other SSH session and issue these commands while positioned in /var/www/html/.well-known/acme-challenge:
mkdir ODDBALL-NAME cd ODDBALL-NAME echo "ODDBALL-RESPONSE > index.html" chown -R asterisk:www-data /var/www/html/.well-known
7. Use a browser to (quickly) go to http://YOUR-FQDN/.well-known/acme-challenge/ODDBALL-NAME/ and be sure your web server displays the expected ODDBALL-RESPONSE. You’ve got to get this working before you continue with the certbot install or it will fail. You only have a few minutes to do this before certbot will change the ODDBALL-NAME and ODDBALL-RESPONSE credentials. 3 consecutive failures and you have to wait an hour to try again. Guess how we know?
8. Once you get the expected response, switch back to your SSH session running the certbot installer and press ENTER to continue with the certificate install. When it completes, you’ll get a congratulatory note and a reminder that, in less than 90 days, you’ll need to run certbot renew to update your certificate.
9. Install the new certificates in NGINX and put things back together again:
cd /etc/nginx/sites-enabled nano -w xivo
10. Begin by removing the 3 # signs that we inserted to get HTTP working in step #2.
11. Near the bottom of the file, comment out these existing certificate lines:
# ssl_certificate /usr/share/xivo-certs/server.crt; # ssl_certificate_key /usr/share/xivo-certs/server.key; # ssl_ciphers ALL:!aNULL:!eNULL:!LOW:!EXP:!RC4:!3DES:!SEED:+HIGH:+MEDIUM;
12. Add the following new lines just below the lines you commented out. Be sure to replace YOUR.FQDN in each line with the actual FQDN of your server:
ssl_certificate /etc/letsencrypt/live/YOUR.FQDN/fullchain.pem; ssl_certificate_key /etc/letsencrypt/live/YOUR.FQDN/privkey.pem; ssl_ciphers HIGH:MEDIUM:!aNULL:!MD5:!SEED:!IDEA;
13. Save the file and then restart your firewall and NGINX:
iptables-restart /etc/init.d/nginx restart
14. Edit /etc/apt/sources.list and comment out the jessie-backports line from step #1.
15. Reload your aptitude sources: apt-get update
16. Remember that you must renew your free certificate every 90 days by issuing this command: certbot renew --quiet
.
Better yet, issue the following command to set up a cron job to auto-renew your certificate every week:
echo "5 3 * * 0 root /usr/bin/certbot renew --quiet > /dev/null 2>&1" >> /etc/crontab
17. Test things out with a web browser by visiting your FQDN. Your browser should now show the site as SECURE.
18. Now try out that new WebRTC phone.
Published: Monday, September 25, 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…
RTPbleed Security Alert: Asterisk Calls Can Be Intercepted
If you’ve installed Asterisk® during the past 4½ years, your server has a MAJOR security problem. If you didn’t already know, with Asterisk, your VoIP conversations actually are carried over a random UDP port using the Real Time Protocol (RTP), not the SIP port (UDP 5060) which handles the setup and teardown of your VoIP connections. It turns out that, since March 2013, all of that RTP traffic and thus your conversations could be intercepted and redirected by anyone on the Internet. As this recent article in The Register noted:
The problem occurs when [communications] systems like IP telephony have to get past network address translation (NAT) firewalls. The traffic has to find its way from the firewall’s public IP address to the internal address of the device or server, and to do that, RTP learns the IP and port addresses to associate with a call.
The problem is, the process doesn’t use any kind of authentication.
This is exacerbated by the fact that, by default, Asterisk and FreePBX® traditionally use the NAT=yes setting (whether needed or not) to enable this navigational magic just in case your calls need it. Without it, you may end up with no audio or one-way audio on your calls. Traditional wisdom was that an attacker needed to be positioned between the caller and the Asterisk server in order to intercept this media stream. As luck would have it, it turns out the man in the middle didn’t need to be in the middle after all. He could be anywhere on the Internet. The old adage to talk on the phone as if someone else were listening turns out to have been pretty good advice in the case of Asterisk communications. Even if you had a firewall, chances are you protected UDP port 5060 while exposing and forwarding UDP 10000-20000 to Asterisk without any safeguards.
According to last week’s Asterisk advisory, “To exploit this issue, an attacker needs to send RTP packets to the Asterisk server on one of the ports allocated to receive RTP. When the target is vulnerable, the RTP proxy responds back to the attacker with RTP packets relayed from the other party. The payload of the RTP packets can then be decoded into audio.” Specifically, if UDP ports 10000-20000 are publicly exposed to the Internet, anybody and everybody can intercept your communications without credentials of any kind. WOW!
So, there’s a patch to fix this, right? Well, not exactly:
Note that as for the time of writing, the official Asterisk fix is vulnerable to a race condition. An attacker may continuously spray an Asterisk server with RTP packets. This allows the attacker to send RTP within those first few packets and still exploit this vulnerability.
The other recommended "solutions" aren’t much better:
- When possible the nat=yes option should be avoided
- To protect against RTP injection, encrypt media streams with SRTP
- Add config option for SIP peers to prioritize RTP packets
The nat=no option doesn’t work if you or your provider employs NAT-based routers. The SRTP option only works on more recent releases of Asterisk, and it also requires SRTP support on every SIP phone. Prioritizing RTP packets is not a task for mere mortals.
Surprisingly, the one solution that is not even mentioned is hardening your firewall to block incoming UDP 10000-20000 traffic that originates outside your server. Our recognized SIP expert on the PIAF Forum had the simple solution. Bill Simon observed:
If the SDP in the INVITE or subsequent re-INVITE contains routable IP addresses, then use them for media. If the SDP contains non-routable IP addresses, then the client is behind a NAT and not using any NAT traversal techniques like SIP ALG, ICE/STUN, so send to the originating IP. Why are we making allowances here for media to come from anywhere? I think you can probably clamp down your firewall as much as you want, because symmetric RTP should allow media to get through by way of establishing an outbound stream (inbound stream comes back on the same path).
Our testing confirms that simply blocking incoming RTP traffic on your firewall solves the problem without any Asterisk patch. In short, RTP traffic cannot originate from anonymous sources on the Internet.
For those using Incredible PBX® or Travelin’ Man 3 or an IPtables firewall, the fix is easy. Simply remove or comment out the INPUT rule that looks like this and restart IPtables:
-A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
On RedHat/CentOS servers, the rule is in /etc/sysconfig/iptables. On Debian/Ubuntu and Raspbian servers, you’ll find the rule in /etc/iptables/rules.v4. On Incredible PBX for Issabel servers, you’ll find the rule in /usr/local/sbin/iptables-custom. On all Incredible PBX platforms, remember to restart IPtables using only this command: iptables-restart.
Published: Friday, September 8, 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…