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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiGame 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…
Another Perfect Pair: Flawless VoIP with Wazo and 3CX
We previously documented how to interconnect an Issabel PBX with 3CX to take advantage of the best of both worlds. Today, we’ll again use the Nerd Vittles free 3CX server offering and interconnect it with a Wazo PBX. An added benefit of using Wazo is the fact that you can set up redundant (and free) HA servers with Wazo in minutes. Once we get the pieces in place, from Wazo extensions, you’ll be able to call your 3CX Clients by dialing 4 digits. And, from 3CX Clients, you can call Wazo extensions as well as all of your Asterisk® applications in the same way with the added bonus of being able to make outbound calls through your Wazo trunks by dialing any number with an 8 prefix from 3CX extensions. Once you have both of your PBXs running, the setup time to interconnect them is under 5 minutes.
Why would you want to maintain two PBXs? As we previously noted, the simple answer is the added flexibility you achieve coupled with a 99% reduction in VoIP headaches. If you haven’t yet used 3CX Clients on a PC or Mac desktop or on an iOS or Android device, you have missed perhaps the greatest VoIP advancement of the last decade. As the name suggests 3CX Clients connect to a 3CX server with less than a one-minute setup. They work flawlessly from anywhere using WiFi or cellular. Every function you’re accustomed to on a top-of-the-line desktop SIP phone works exactly the same on the 3CX clients: phonebook, hold, transfer, voicemail, chat, conferencing, and WebMeeting. It’s what every Unified Communications system should deliver. The silver lining is you can kiss all of your Asterisk NAT woes goodbye! If you ever travel or if you need remote phone access to your PBX infrastructure, you owe it to yourself to try a 3CX Client. We promise. You’ll never go back!
Building Your Wazo and 3CX Server Platforms
The prerequisite for interconnecting Wazo and 3CX servers is, of course, to install the two PBXs on platforms of your choice. Our preference is cloud-based servers because it avoids many of the stumbling blocks with NAT-based routers. If you know what you’re doing, you obviously can deploy the PBXs in any way you like. For the Wazo PBX, start with our latest Wazo tutorial. For 3CX, start with our introductory tutorial which includes a link to obtain a free perpetual license supporting 4 simultaneous calls and unlimited trunks. Then secure your server by adding the Travelin’ Man 3 firewall for 3CX. Once both servers are up and running, whitelist the IP address or FQDN of the Wazo PBX on the 3CX server and vice versa. You’ll find the add-ip and add-fqdn utilities in /root of each server.
Overview of Interconnection Methodology
If you’re new to all of this, suffice it to say that 3CX is a powerful, commercial PBX while Wazo provides a robust Asterisk RealTime implementation for basic telephony operation. The two systems are quite different in terms of their approaches to interconnectivity. While you can transparently interconnect one 3CX server to another one, you cannot accomplish the same thing when the second PBX is Asterisk-based. Instead, Wazo is configured as a SIP trunk on the 3CX platform. The limitation this causes is that extensions on the Wazo PBX can only direct dial extensions on the 3CX platform. Wazo-based extensions cannot utilize 3CX trunks to place outbound calls. There’s more flexibility on the 3CX side of things. 3CX extensions can place direct calls to Wazo extensions. They also can take advantage of Wazo’s trunks to place outbound calls. Additionally, as we noted above, 3CX extensions can take advantage of every Asterisk application hosted on the Wazo platform including all of the Incredible PBX® enhancements. This actually works out perfectly because you can deploy 3CX Clients for your end-users, and they can take advantage of all the extension and trunk resources on both the 3CX and Wazo platforms. It also greatly simplifies remote deployment by removing NAT one-way audio hassles while allowing almost instantaneous setup of remote 3CX Clients, even by end-users.
For our setup today, we’re assuming you have elected to use 3-digit extensions on both the Wazo and 3CX platforms. To call extensions connected directly to the alternate server, we will simply dial 8 + the extension number on the remote PBX. To make external calls from 3CX extensions using Wazo trunks, we will dial 8 + a 10-digit number. For international users, you can adjust the dialplan on both PBXs accordingly.
By default, SIP trunks are associated with a DID on the 3CX platform. We will register the 3CX DID trunk with Wazo to maintain connectivity; however, we will not register the corresponding trunk on the Wazo side with the 3CX server. Keep in mind that you can only route a 3CX DID to a single destination, i.e. an extension, a ring group, or an IVR. But we can use 3CX’s CallerID routing feature to send calls to specific 3CX extensions from Wazo extensions even using a single 3CX trunk. For each 3CX extension, we’ll create an Outbound Route on the Wazo side with a CallerID number that matches the 3CX extension number we wish to reach. On the 3CX side, we’ll create an Inbound CID Rule that specifies the extension number to which each matching CallerID number should be routed. This sounds harder than it actually is. So keep reading, and it’ll all make sense momentarily. Once you’ve set all of this up, we think you’ll agree that it makes sense to create the bulk of your extensions exclusively on the 3CX side.
Configuring Wazo for Interconnection to 3CX
Let’s begin by creating a Trunk on the Wazo side to connect to your 3CX server. In the Wazo GUI, choose IPBX:Trunk Management:SIP Protocol and + Add SIP Trunk.
In the General tab, fill in the blanks as shown below. Make up a very secure Password:
In the Signalling tab, fill in the blanks identified by arrows as shown below:
In the Advanced tab, fill in the blanks as shown below. Then SAVE the trunk settings.
Because we set up the Wazo trunk with a Default destination context, we don’t need an Incoming Route for the 3CX calls since they will be processed exactly as if they were dialed from a local extension on the Wazo PBX, i.e. local calls will be routed to extensions and outgoing calls through trunks will be routed using your existing Outbound Routes.
Finally, we need to create the Outbound Routes for calls originating from Wazo extensions that should be directed to specific extensions on the 3CX platform. You’ll need a list of the 3CX extension numbers you wish to enable on the Wazo platform, and we’ll need to create a separate Outbound Route for each 3CX extension to be enabled. Create the Outbound Routes using the template below after accessing Call Management:Outgoing Calls:+ Add Route.
In the General tab, we recommend including the 3CX extension in the Name field. The Context should be Outcalls, and the Trunk should be the 3CX001 trunk we created above.
In the Exten tab, specify the dialing prefix (9) followed by the 3CX extension number in the Exten field. Then choose 1 in the Stripnum field to tell Wazo to strip off the dialing prefix before sending the call to the 3CX PBX. Click SAVE to save your new outbound route settings. Repeat for each 3CX extension that should be accessible from the Wazo PBX.
Configuring 3CX for Interconnection to Issabel PBX
Now we’re ready to set up the 3CX side to interconnect with your Wazo PBX. Start by creating a SIP Trunk and fill out the template as shown below using one of the phone numbers associated with your Wazo PBX as the Main Trunk No.
Fill in the Trunk Details using the example below. Be sure to specify the actual IP address or FQDN of your Wazo server as well as the SIP credentials of 3CX for username and the actual password you set up on the Wazo side of things. The Main Trunk No will be the same as you entered in the previous step. Choose a Default Destination for the Trunk.
When the SIP Trunks listing redisplays, highlight your new Asterisk trunk and click Refresh Registration. The icon beside the Trunk should turn green. If not, be sure your IP address and password match the settings on the Wazo side. Remember to also whitelist the IP address of your 3CX server on the Wazo PBX using /root/add-ip and do the same for the Wazo PBX on the 3CX side. Don’t proceed until you get a green light!
Now we need two Outbound Routes for calls placed from 3CX extensions. One will handle calls destined for Local Extensions on the Wazo side. Our design is to place calls to Wazo extensions by dialing 8 + the 3-digit extension number. Adjust this to meet your own requirements. Be sure to set the Route as Wazo with a value of 1 for Strip Digits.
The other Outbound Route will handle calls destined for external calling with a Wazo trunk using a similar methodology. 3CX users will dial 8 + 10-digit number for calls to be processed by Trunks on the Wazo server.
Finally, we need an Inbound Rule for every 3CX extension that you wish to enable for remote calling from Wazo extensions. Use the Add CID Rule option to create each Inbound Rule using the sample below. In our example, we’re authorizing incoming calls to 3CX extension 003 where the CallerID number of the incoming call is 003. This template is exactly the same as what we used with the 3CX-Issabel setup previously.
Test Drive Your Interconnected Servers
Now we’re ready to try things out. From an extension on the 3CX server, dial 8 plus any 3-digit extension that exists on the Wazo server. Next, dial 8 plus a 10-digit number such as your smartphone. The call should be routed out of your Wazo server using the Trunk associated with the NXXNXXXXXX rule in your Wazo Outbound Routes. Finally, from an extension on your Wazo PBX, dial 9 plus 000 which should route the call to extension 000 on your 3CX server. Enjoy!
Published: Tuesday, September 5, 2017
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Finding the Perfect Phone Solution for Small Organizations
Many of us wear several hats during our business careers. One of those invariably is managing a community organization of some flavor. We frequently are asked for advice on what the ideal telephony solution would be for such an organization. The reason for the inquiries typically is because the Bell Sisters have now jacked up the cost of a single, business phone line to well over $100 a month. And that gets you local calls only unless you sign up for exorbitant additional charges for long distance calling. It’s worth noting that most of the individuals making these inquiries stress that they do not want to get in the business of managing a phone system. They’re looking for a plug-and-play, set-it-and-forget-it setup that will require minimal tweaking. My first question is always: "What’s your budget?" Then we explore (1) how many phones, (2) the frequency of calls, (3) the number of simultaneous calls, (4) the mix of local and long distance calling, and, last but not least, (5) the must-have feature set. No shocker: the budget is always near zero.
Today, we’re going to start on the bottom rung and work our way up the technology ladder. If you never thought smartphones and cellular would be part of this equation, guess again. $60 will now buy you a 4G LTE smartphone at WalMart, and monthly plans with unlimited calling in the U.S. start at $25 for Walmart’s Family Mobile plan, a far cry from the Ma Bell business phone rates. And you can keep your number! If you need multiple phones but only a single line, that’s not a problem either. Add a Link2Cell digital cordless phone system from Panasonic and now you have as many as 5 phones that can make and receive calls using your cellular connection via Bluetooth®. Some even support a second cellphone connection. With many you can build a phonebook on your cellphone and import it into all of your cordless phones. And, of course, voicemail is included as part of your cell plan. For those with poor cellular service, the Family Calling Plan supports free WiFi calling on many cellphones. And $10 extra buys you rollover international calling funds with 5¢/min. rates to Canada and Mexico. Calling rates to other countries are less than impressive and do not compare favorably with typical VoIP rates.
Cellular phone service isn’t for everyone, and there are considerably more choices in the Land of VoIP. The wrinkle with all of the VoIP solutions is that now you need internet service at the site of your organization. To say there is minimal competition in the internet service provider market is an understatement. If you’re lucky, you’ll have a choice between AT&T and one of the cable companies: Comcast, Charter, or Time Warner/Spectrum. The downside is it adds an additional $25 to $75+ to your monthly costs unless the organization already has Internet service that is used for purposes other than telephony. What won’t work for VoIP is satellite internet service because of latency issues.
Once you’re over the internet service hurdle, there are numerous VoIP choices for phone service depending upon your skillset. Again, let’s start on the bottom rung. If you can make it with one phone and one call at a time, it’s hard to beat Ooma Telo. $100 buys you a device that delivers landline-like phone service at a monthly cost of $4 (you only pay communications taxes and fees) to $10 depending upon the feature set you choose. The basic, fees-only plan gets you toll-free nationwide calling in the U.S., call waiting, caller ID, 911 service, a call log history and voicemail through Ooma’s online dashboard. The premium $10 a month plan adds a second line, free calling to Canada and Mexico, voicemail via email, call screening, do not disturb and call forwarding to an Android phone or iPhone. As with cellular service, you can keep your existing phone number. If you need WiFi connectivity or cellphone Bluetooth connectivity for your Ooma device, add $50. Otherwise, just plug a standard telephone into the Ooma hardware, and you’re good to go. You also could use a wireless phone system such as the ones described in the previous section to add up to five extensions.
If you need additional lines or phones, the $200 Ooma Office offering is worth considering. You can add as many users as desired for $19.95/month/each with every user getting unlimited U.S./Canada calling, CallerID service, and an impressive collection of business phone features (shown above). The cost of the VoIP phones for each user are not included. While the monthly service charges are pricey, you’re paying for the simplicity of never having to deal with the intricacies of configuring and managing a business phone system. However, you do have to purchase and configure a SIP phone for each user.
When you get beyond the single user, single line requirement, the sky opens up in the VoIP market. The savings go from getting part of your hundred dollars back each month to saving several hundred or thousands of dollars every month. What becomes important is how much of the deployment work you’re willing to undertake yourself. If the answer is not much, then the phone systems from one of our corporate sponsors, 3CX or RentPBX, are probably your best bets. Both offer turnkey VoIP solutions, and 3CX also has a worldwide dealer network to handle all of the deployment chores for you as well. While the front end costs with the 3CX commercial solution must be considered, the long-term savings more than cover these costs in your first year.
If you’re capable of making your own dinner by reading the directions off the side of a box, then you can probably handle many VoIP deployments yourself. The list of tasks goes something like this. You’ll either need a computer or cloud provider for a computing platform. Then you need a Linux operating system for that platform. Next, you need VoIP software to serve as your PBX. Services such as RentPBX handle setup of all three of these tasks for a monthly cost of $15. Or you can do it yourself and reduce the cost to $5 or less per month. We have dozens of tutorials to show you how.
At this juncture, you’re pretty much on your own except for our tutorials. The remaining tasks include purchasing and configuring phones for your users and configuring trunks from one or more VoIP providers, the folks that interconnect your phone calls to the people you are calling. Then you configure your PBX to route calls in and out of your PBX, and you’re in business. All of these tasks are managed using web-based GUI software, and there are plenty of tutorials to hold your hand every step of the way.
We’ll finish up today by walking you through one of our favorite open source VOIP setups. It provides free calling and faxing in the United States. Typical setup takes less than an hour, and the monthly cost is $3 which includes nightly backups of your entire PBX. These backups can be restored with a single button click.
FULL DISCLOSURE: 3CX, RentPBX, Amazon, Vitelity, and Vultr all provide financial support to Nerd Vittles and our open source projects. We’ve chosen these providers not the other way around. Our decisions were based upon their corporate reputation and the quality of their offerings and their pricing,
The Vultr/VoIP Open Source Solution
Begin by setting up an account at Vultr using our referral link. Then create a new instance choosing the smallest Server Size and CentOS 7/64-bit as the Server Type. Pick a Server Location that supports the $2.50 server size. Currently, Miami and New York are available. Once your virtual machine is running, you can activate automatic backups under the Server Information:Backups tab in the Vultr Control Panel.
(1) Once you’ve built and started your new virtual machine, log into your server as root using SSH/Putty and immediately change your root password: passwd.
(2) With the $2.50 size VULTR virtual machine, you must create a swapfile before proceeding. Here are the commands:
dd if=/dev/zero of=/swapfile bs=1024 count=1024k chown root:root /swapfile chmod 0600 /swapfile mkswap /swapfile swapon /swapfile echo "/swapfile swap swap defaults 0 0">>/etc/fstab sysctl vm.swappiness=10 echo vm.swappiness=10>>/etc/sysctl.conf free -h cat /proc/sys/vm/swappiness
(3) Now you’re ready to kick off the Issabel 4 install. Here are the commands:
cd /root yum -y install wget nano dialog wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash
When prompted for a MySQL password, use: passw0rd (with a zero). Choose a secure Issabel admin password for the GUI.
(4) After the reboot, log back in as root and install Incredible PBX for Issabel:
cd /root wget http://incrediblepbx.com/IncrediblePBX11-Issabel4.sh chmod +x IncrediblePBX11-Issabel4.sh ./IncrediblePBX11-Issabel4.sh
When prompted for a MySQL password, use: passw0rd (with a zero). Choose a secure Issabel admin password for the GUI.
(5) After the reboot, configure your correct timezone: /root/timezone-setup
Be advised that, when you log into the Issabel web interface, you will be prompted (three times) for your admin credentials. You can save these entries to avoid having to repeat it in the future. Now you can jump over to the Incredible PBX for Issabel tutorial to complete your installation. Within a couple minutes, your PBX will be ready to accept calls. Enjoy!
Published: Monday, August 7, 2017
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
Almost Free: Professional Grade TTS Comes to Issabel 4
There’s no need to be chained to your TV for breaking news and weather forecasts when they can be as close as the nearest VoIP phone. Today we’re elevating text to speech with Issabel to commercial-quality. We’re wrapping up our month-long romance with Issabel 4 by introducing IBM’s Bluemix TTS service for Incredible PBX®. It’s surprisingly affordable. The first million characters of text-to-speech synthesis are FREE every month so, for most users, upgrading to commercial quality speech synthesis is a no-brainer. Try out our 10-second demo and prepare to be amazed. We provided a plain text demo (without any voice transformation SSML) to show how incredibly accurate IBM’s basic voice synthesis engine is. With additional tweaking using IBM’s SSML functions, any voice nuances can be quickly corrected or enhanced. Feel free to build a few samples on your own at IBM’s demo site.
[soundcloud url="https://api.soundcloud.com/tracks/335398310″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="80%" height="414″ iframe="true" /]
An awesome text-to-speech engine, of course, is only half of the story. You still need application software to bring TTS to life on your PBX. Nerd Vittles tried and true news and weather applications for Incredible PBX provide the glue that binds news and weather updates to your phone by simply dialing a 3-digit extension on your PBX. 951 gets you the latest breaking news from Yahoo, and 947 gets you current weather conditions and a weather forecast for any zip code in the United States. It’s pure, open source GPL code so feel free to modify it to meet your needs. Additional weather data is available from IBM Bluemix at modest cost for our international friends. Make that your weekend project!
Getting Started with IBM Bluemix TTS Service
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
You can start your free, 30-day trial of IBM Bluemix services without providing a credit card. Just sign up here. Once your account is activated, here’s how to obtain credentials for the TTS service to use with Incredible PBX for Issabel. Start by logging in to your IBM Bluemix account. Once you’re logged in, click on your account name (1) in the upper right corner of your web page to reveal the pull-down to select your Region, Organization, and Space. Follow the blue links at the bottom of the pull-down menu to create an Organization and Space for your TTS service.
Next, click the Menu icon which is displayed as three horizontal bars on the left side of the web page. Choose Watson. Click Create Watson Service and select Text to Speech from the applications listing. Watson will generate a new TTS service template and display it. Make certain that your Region, Organization, and Space are shown correctly. Then verify that the Standard Pricing Plan is selected. When everything is correct, click the Create button.
When your Text to Speech application displays, click Service Credentials and then click New Credential (+). When the Add New Credential dialog appears, leave the default settings as they are and click Add. Your Credentials Listing then will appear. Click View Credentials beside the new entry you just created. Write down your URL, username, and password. You’ll need these to configure the IBM Bluemix TTS service in Issabel momentarily. Logout of the IBM Cloud by clicking on the little face in the upper right corner of your browser window and choose Log Out. Confirm that you do, indeed, wish to log out. NOTE: For new implementations, you will have an APIkey instead of a username and password.
Implementing IBM Bluemix TTS Service with Issabel
Now for the fun part. We’ve built all the pieces you’ll need to deploy IBM’s TTS service and to reconfigure the Incredible PBX news and weather applications to take advantage of IBM’s new text synthesis engine. There are 5 Simple Steps to put all the pieces in place for this. Begin by (1) installing Issabel 4 on your favorite platform. Next, (2) install Incredible PBX for Issabel by following our tutorial. Now (3) log into your Issabel PBX as root using SSH or Putty and issue the following commands:
cd /var/lib/asterisk/agi-bin wget http://incrediblepbx.com/ibmtts-issabel.tar.gz tar zxvf ibmtts-issabel.tar.gz nano -w /var/lib/asterisk/agi-bin/ibmtts.php
When the installation finishes, (4) an editor will open to let you insert your IBM Bluemix TTS credentials. Do so and then press Ctrl-X, Y, and Enter to save your entries. For new deployments, your API Username will be apikey, and your API Password will be your actual APIkey. Finally, while still in the agi-bin directory, (5) run the following script to update your Asterisk dialplan: ./install-ibmtts-dialplan.sh.
Now you’re ready to take IBM’s Bluemix TTS for a test drive. Pick up any phone connected to your PBX and dial 951 for the latest Yahoo news. Then dial 947 and enter a 5-digit zip code to retrieve the latest weather conditions and weather forecast for your zip code. Enjoy!
If you’d like to try out the News application with IBM Bluemix, feel free call our Demo PBX and choose option 5:
Published: Monday, July 31, 2017
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
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 Perfect Pair: Flawless VoIP with Issabel 4 and 3CX
We continue our Issabel 4 adventure today with a VoIP match made in heaven. Today, we’ll take advantage of the Nerd Vittles free 3CX server offering and interconnect it with an Issabel 4 PBX to enjoy the best of both worlds. From Issabel extensions, you can call your 3CX Clients by dialing 4 digits. From 3CX Clients, you can call Issabel extensions as well as your Asterisk® applications in the same way with the added bonus of being able to make outbound calls through your Issabel trunks by dialing any number with a 9 prefix. 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? 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. As an added bonus, 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 again use a traditional SIP client.
Building Your Issabel and 3CX Server Platforms
The prerequisite for interconnecting Issabel 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 Issabel 4 PBX, start with our introductory tutorial to install Issabel 4. Then follow the Incredible PBX for Issabel tutorial to add security and the Asterisk bells and whistles. 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 Issabel 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 Issabel relies upon Asterisk and FreePBX® for its 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, the Issabel PBX is configured as a SIP trunk on the 3CX platform. The limitation this causes is that extensions on the Issabel PBX can only direct dial extensions on the 3CX platform. Issabel-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 Issabel extensions. They also can take advantage of Issabel-based trunks to place outbound calls. Additionally, as we noted above, 3CX extensions can take advantage of every Asterisk application hosted on the Issabel 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 Issabel 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 Issabel and 3CX platforms. To call extensions connected directly to the alternate server, we will simply dial 9 + the extension number on the remote PBX. To make external calls from 3CX extensions using Issabel trunks, we will dial 9 + 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 Issabel to maintain connectivity; however, we will not register the corresponding trunk on the Issabel 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 Issabel extensions even using a single 3CX trunk. For each 3CX extension, we’ll create an Outbound Route on the Issabel 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 Issabel PBX for Interconnection to 3CX
Let’s begin by creating a Trunk on the Issabel PBX to connect to your 3CX server. In the Issabel GUI, choose PBX:PBX Config:Trunks and Add a SIP Trunk. Fill in the blanks as shown below. Make up a very secure secret for your Trunk and be sure to leave the Outbound CallerID field blank. Click on the image below if you need to enlarge it.
Because we set up the 3CX trunk with a from-internal destination context, we don’t need an Incoming Route for the 3CX Trunk. The calls will be processed exactly as if they were dialed from a local extension on the Issabel 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 Issabel 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 Issabel 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. We recommend including the 3CX extension in the Route Name. The Route CID and Route Pattern should be a 9 followed by the 3CX extension number for each Outbound Route you create. In the template below, we’re telling Issabel to route a call dialed as 9003 to extension 003 on the 3CX PBX. The Dial Manipulation Rule in the 3CX Trunk settings tells Issabel to strip off the 9 before sending the call to the 3CX PBX.
Configuring 3CX for Interconnection to Issabel PBX
Now we’re ready to set up the 3CX side to interconnect with your Issabel PBX. Start by creating a SIP Trunk and fill out the template as shown below using one of the phone numbers associated with your Issabel 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 Issabel server as well as the SIP credentials of 3CX for username and the actual password you set up on the Issabel 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 Issabel 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 Issabel side. Our design is to place calls to Issabel extensions by dialing 9 + the 3-digit extension number. Adjust this to meet your own requirements. Be sure to set the Route as Asterisk with a value of 1 for Strip Digits.
The other Outbound Route will handle calls destined for external calling with an Issabel trunk using a similar methodology. 3CX users will dial 9 + 10-digit number for calls to be processed by Trunks on the Issabel server.
Finally, we need an Inbound Rule for every 3CX extension that you wish to enable for remote calling from Issabel 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.
Test Drive Your Interconnected Servers
Now we’re ready to try things out. From an extension on the 3CX server, dial 9 plus any 3-digit extension that exists on the Issabel server. Next, dial 9 plus a 10-digit number such as your smartphone. The call should be routed out of your Issabel server using the Trunk associated with the NXXNXXXXXX rule in your Issabel Outbound Routes. Finally, from an extension on your Issabel PBX, dial 9 plus 000 which should route the call to extension 000 on your 3CX server. Enjoy!
Published: Wednesday, July 19, 2017
Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a difficult place to address support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forum. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Unlike some forums, the PIAF Forum is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of users just like you. You won’t have to wait long for an answer to your question.
Need help with Asterisk? Visit the PBX in a Flash Forum.
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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…
Leap Into Summer: Introducing Incredible PBX for Issabel
NEWS FLASH: A new release of Incredible PBX for Issabel is now available. Tutorial is here.
If you didn’t already know, we’ve always liked free. No strings, no gotchas, no demoware, and no legal shenanigans. That’s why our introduction of Issabel 4 last week was such a breath of fresh air. While there’s now an awesome free version of 3CX, the open source community has had a very long dry spell. So today we celebrate a decade of adding fun to phone systems with the introduction of Incredible PBX® for Issabel 4. It includes our next generation, preconfigured Travelin’ Man 3 firewall, additional text-to-speech engines (FLITE, GoogleTTS, and PicoTTS), voice recognition, turnkey trunk and extension setups with preconfigured tollfree calling, Google Voice support with OAuth 2 or plain text passwords for free calling in the U.S. and Canada, SMS messaging, telephone reminders, turnkey fax support, AsteriDex phone book with both voice and speed dialing, Wolfram Alpha, sample ODBC apps, and a boatload of dialplan code and AGI scripts to help anyone wanting to learn how to develop custom applications with Asterisk®.
Installing Incredible PBX for Issabel 4
Let’s start with the basics and get all of the Incredible PBX components loaded. As with all Incredible PBX builds, running the Incredible PBX installer will erase ALL of your existing Issabel configuration. So begin with a clean, unaltered Issabel 4 platform with no added components or configuration changes. Be sure to use either the June or July ISO for base Issabel install. We will update it from there as part of the Incredible PBX install. Just follow last week’s tutorial to bring up Issabel 4 on a dedicated server or a virtual machine.
JUST RELEASED: A new tutorial to walk you through Getting Started: Issabel in the Cloud.
The Travelin’ Man 3 firewall is installed and configured as part of the install. It whitelists certain IP addresses and blocks everyone else from even seeing your server on the Internet. For this reason, it is critically important that you perform the Incredible PBX install using SSH or Putty from a PC that you will use to manage your Issabel server. Otherwise, you risk locking yourself out of your own server. Whitelisted IP addresses include the Issabel server itself, the public and private IP addresses of your desktop PC, all non-routable, private LAN addresses, and the Nerd Vittles collection of recommended SIP hosting providers. You can add as many additional providers or users to the whitelist using the simple tools provided as part of the install and further documented below. Do NOT activate Issabel’s firewall.
As part of the install process, you’ll be prompted during both passes to create a password for MySQL/MariaDB and an admin password for the Issabel web GUI. The MySQL password MUST be passw0rd (with a zero), or you will get a permanent mess. The admin password can be anything you like. Passwords can be updated by running /root/admin-pw-change. Many of the Incredible PBX apps depend upon this MySQL password so don’t change it. Your MySQL databases remain secure and can only be accessed on localhost or after a successful root login to your server from a whitelisted IP address.
Begin the Incredible PBX install by logging into your Issabel server as root from a desktop PC using SSH or Putty and execute the following commands:
cd /root wget http://incrediblepbx.com/IncrediblePBX11-Issabel4.sh chmod +x IncrediblePBX11-Issabel4.sh ./IncrediblePBX11-Issabel4.sh
Introducing the (new) Travelin’ Man 3 Firewall
Issabel 4 includes an IPtables firewall component. Do NOT activate it because Incredible PBX includes its own preconfigured IPtables firewall, better known as Travelin’ Man 3. With the Issabel 4 firewall, the administrator is responsible for setting all of the firewall rules. With Travelin’ Man 3, all the heavy lifting is done for you. The design is also markedly different. Issabel 4 opens ports which you define, but it gives worldwide access to those ports by any user. Travelin’ Man 3 employs a WhiteList rather than opening ports for everyone. If you’re on the WhiteList, you get access to the limited collection of ports assigned to that IP address. If you’re not on the WhiteList, you cannot even see the Issabel PBX from the Internet. For those without remote telephones or traveling employees, this provides total protection of your server with virtually no further firewall management.
If you have remote users of your PBX or if you wish to deploy softphones on mobile devices and rely upon WiFi facilities at random locations, Travelin’ Man 3 provides several utilities to assist. If the remote users have static IP addresses, then those IP addresses can be added to the WhiteList by running /root/add-ip. Better yet, a NeoRouter VPN is provided that lets remote users access Issabel using NeoRouter private LAN addresses that already are WhiteListed as part of the installation process. These require little to no configuration with static or dynamic IP addresses even when switching between WiFi networks. For those with dynamic IP addresses and no VPN, FQDNs can be assigned using a service such as dyn.com and a dynamic DNS client can be loaded on the smartphone to keep the current IP address synchronized with the FQDN. On the Incredible PBX side, these FQDNs can be added using /root/add-fqdn, and the IP addresses will be updated automatically every 10 minutes. The final option to provide remote users the 3-digit PortKnocker codes from knock.FAQ and let them automatically whitelist their own IP addresses by running the PortKnocker client from any smartphone or Linux server. When the Issabel server detects a successful knock sequence, the source IP of the knock sequence is whitelisted until the next reload of the firewall. If an administrator prefers to allow permanent additions to the WhiteList that survive a reboot or restart of the firewall, the administrator need only run the following command one time: iptables-knock activate. WhiteListed entries can be removed using the /root/del-acct utility. Further details on the new Travelin’ Man 3 design are available here.
Update: The July Issabel ISO introduced a quirk into our Travelin’ Man 3 setup. For a reason that we have not yet tracked down, it is no longer possible to whitelist an IP address and use that address to access the Issabel GUI with a browser. Until we can track down the problem, we have modified the security methodology to access the Issabel web GUI. While we have opened port 443 for public access, we have added another layer of security by requiring Apache htaccess credentials before you can access any web site on your Issabel server. As the last step of the Incredible PBX installation procedure, you will be prompted to enter your admin password again. The username admin and the admin password are used BOTH for Apache authentication AND Issabel GUI authentication. Should you ever need to change your Issabel GUI admin password using /root/admin-pw-change, you also will need to execute the following command to change the admin password for Apache authentication: htpasswd -c /etc/pbx/wwwpasswd admin.
Setting Up a Softphone with Issabel 4
If you’re a Mac user, you’re lucky (and smart). Download and install Telephone from the Mac App Store. Start up the application and choose Telephone:Preference:Accounts. Click on the + icon to add a new account. To set up your softphone, you need 3 pieces of information: the IP address of your server (Domain), and your Username and Password. You can decipher your server’s IP address by running pbxstatus. If you wish to use one of the preconfigured extensions (701 and 702), you’ll find the randomized passwords in /root/passwords.FAQ. Now copy or cut-and-paste your Username and Password into the Accounts dialog of the Telephone app. Click Done when you’re finished, and your new softphone will come to life and should show Available. Dial the IVR (D-E-M-O) to try things out. With Telephone, you can use over two dozen soft phones simultaneously.
For everyone else, we recommend the YateClient softphone which is free. Download it from here. Run YateClient once you’ve installed it and enter the credentials for the Issabel extension. You’ll need the IP address of your server plus your extension number and password associated with either the 701 or 702 extension.
Configuring Google Voice Natively or Using Simonics
Everybody likes free calling, and nobody does it better than Google. Will it last? Well, the naysayers (including me) have been predicting its demise for over 5 years. Yet it keeps on ticking. If you live in the U.S. and want to take advantage of free calls in the U.S. and Canada, you’d be crazy not to deploy a Google Voice trunk on your PBX. Voice quality is near perfect. And the price is right.
The original release of Incredible PBX for Issabel did not support Google Voice trunks so we suggested an intermediary to provide the functionality through a SIP gateway. It works flawlessly using OAuth 2 password authentication, but it’ll set you back $5. If you prefer free, we’ve added the original Google Voice plain-text password solution from the FreePBX® 2.11 days in the latest Incredible PBX release using the July Issabel ISO. It is far from perfect. While you can make and receive calls and faxes to and from Issabel extensions, you cannot direct incoming calls to an IVR because of an old NAT quirk in Asterisk 11. If this isn’t a problem for you, keep reading. Otherwise, skip down to the Simonics tutorial below after completing the initial Google Voice setup which follows.
Here are the initial setup steps on the Google side:
1. Set up a dedicated Gmail and Google Voice account to use exclusively for this new SIP gateway. Head over to the Google Voice site and register. You’ll need to provide a U.S. phone number to verify your account by either text message or phone call.
2. Once you have verified your account by entering your verification code, you’ll get a welcome message from Mr. Google. Click Continue to Google Voice.
3. Provide an existing U.S. phone number for verification. It can be the same one you used to set up your Google account in step #1.
4. Once your phone number has been verified, choose a DID in the area code of your choice.
5. When your DID has been assigned, click the More icon at the bottom of the left column of the Google Voice desktop. Click Legacy Google Voice. Now click the Settings icon on your legacy Google Voice desktop. Make certain that Forward Calls to Google chat is checked and disable calls to your forwarding number. Click on the Calls tab and select Call Screening:OFF, CallerID (Incoming):Display Caller’s Number, and Global Spam Filtering:checked. The remaining entries should be blank.
6. Google Voice configuration is now complete. Sign out of your Google Voice account.
The Native Google Voice Solution using FreePBX Motif Module. Here’s a quick thumbnail of the steps to put all the pieces in place using the FreePBX Google Voice/Motif module. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up the Google Voice account in the Issabel GUI to activate the Google Voice trunk. Next, we’ll add an Incoming Route to tell Issabel how to process Google Voice calls. Then we need to tell Google to relax the rules on use of plain text passwords. And, finally, we’ll restart Asterisk from the Linux CLI.
1. Login to the Issabel web interface with your admin password and choose PBX:PBX Config:Google Voice. Enter your Google Voice account name, password, and 10-digit phone number. Be sure to check all three boxes to Add a Trunk, Add an Outbound Route, and Send Unanswered Calls to Google Voicemail. Click Submit and then reload your dialplan when prompted.
2. Configure an Inbound Route for your incoming Google Voice calls. Click Inbound Routes in the PBX Configuration Menu. Then click Add Incoming Route and enter a Description for the route and enter the DID Number using your 10-digit Google Voice number. If you want to activate CNAM (CallerID Name) lookups, choose OpenCNAM from the Source list. Choose an appropriate Destination for the calls from the pull-down menu of choices. Use only an extension or a ring group. Then SAVE your settings and reload dialplan. To activate fax detection, change Detect Faxes to YES, Detection type to SIP, Detection time to 4, and Destination to Extension 329 (F-A-X). Click Submit and then reload your dialplan again.
3. On the Google site, login into your Google Voice account again. Then follow this link to Enable Less Secure Apps. Then follow this link to activate the Google Voice Reset Procedure. Now log out of your Google Voice account.
4. Login to your Issabel server with SSH/Putty as root and restart Asterisk: amportal restart
5. Now connect a SIP phone to extension 701 and place a call to any number in the U.S. or Canada.
6. Once you have placed an outbound call, incoming calls should work by dialing your Google Voice number from any phone. If you have trouble getting Google to answer the call, this is fairly typical. Try adjusting the NAT settings for your extension from YES to NEVER and place another call. Then change then back to NAT = yes, and you should be good to go.
7. For additional Google Voice trunks, rinse and repeat.
The Simonics GV-SIP Gateway Solution. Here’s the quick thumbnail of the steps to put all the pieces in place. First, we set up a Google Voice account at Google as documented above. Next, we’ll set up an account at the Simonics site to link our Google Voice account to the Simonics SIP Gateway. Then we’ll plug our Simonics SIP credentials into the preconfigured Simonics trunk on Incredible PBX. Finally, we’ll add Incoming and Outgoing Routes to tell Issabel how to process Google Voice calls.
Now you’re ready to set up an account on the Simonics site. With this Nerd Vittles link, there’s a one-time fee of $4.99.
1. Start by registering your new Google account.
2. After paying the $4.99 registration fee via PayPal, proceed through the setup process to link your Google Voice account and 11-digit Google Voice phone number to the Simonics SIP Gateway.
3. You then will be provided your SIP username and password as well as the gateway address, gvgw.simonics.com, to use in building your SIP trunk on your Issabel PBX.
4. If your SIP credentials ever get compromised, regenerate your password by logging back into the Simonics GW site.
Now it’s time to configure your Simonics trunk in Incredible PBX for Issabel. Start by logging into the Issabel web interface as admin with your admin password from above. Next, click PBX:PBX Configuration in the left Issabel menu. Click Trunks:Simonics-GV in the PBX Configuration menu. The Simonics-GV trunk template will display:
1. Untick the Disable Trunk check box.
2. In Outbound CallerID, insert your 10-digit Google Voice number.
3. In username, insert GV1 followed by your 10-digit Google Voice number.
4. In secret, insert your Simonics SIP password.
5. In the Registration String, insert GV1 followed by your 10-digit Google Voice number followed by a colon (:)
6. In the Registration String after the colon, insert your Simonics SIP password.
7. In the tail of the Registration String after the slash (/), insert your 10-digit Google Voice number.
8. Click Submit Changes and then Reload the Dialplan when prompted.
Now you’re ready to configure an Outbound Route for your Google Voice calls. Click Outbound Routes in the PBX Configuration Menu. Then click Add Route and fill out the form as shown below, save your settings, and reload the dialplan.
Finally, let’s configure an Inbound Route for your incoming Google Voice calls. Click Inbound Routes in the PBX Configuration Menu. Then click Add Incoming Route and enter a Description for the route and enter the DID Number using your 10-digit Google Voice number. If you want to activate CNAM (CallerID Name) lookups, choose OpenCNAM from the Source list. Choose an appropriate Destination for the calls from the pull-down menu of choices, e.g. extension, ring group, IVR, etc. Then SAVE your settings and reload dialplan.
Your Google Voice trunk through the Simonics SIP Gateway should now be working. You can verify this by entering sip show registry in the Asterisk CLI. Place a test call from a softphone connected to your Issabel PBX by dialing a 10-digit number. Then place a call to your Google Voice number from a mobile phone or home/office phone. The Asterisk CLI displays progress of calls by activating it from Linux CLI: asterisk -rvvvvvvvvvv
If you have trouble getting Google Voice to work (especially if you have previously used your Google Voice account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
If you want to display your primary phone number on the pbxstatus dialog, simply enter the number in /etc/pbx/.phone.
Adding Speech Recognition Support to Incredible PBX
To support many of our applications, Incredible PBX has included Google’s speech recognition service. These applications include AsteriDex Voice Dialing by Name (411) and Wolfram Alpha for Asterisk (4747), all of which use Lefteris Zafiris’ terrific speech-recog AGI script. Unfortunately (for some), Google now has tightened up the terms of use for their free speech recognition service. Now you can only use it for "personal and development use." If you meet those criteria, keep reading. Here’s how to activate speech recognition on Incredible PBX. Don’t skip any steps!
If you like Siri, you’ll love Wolfram Alpha. To use Wolfram Alpha by phone, you first must obtain a free Wolfram Alpha APP-ID. Then issue the following command replacing APP-ID with your actual ID. Don’t change the yourID portion of the command:
sed -i "s|yourID|APP-ID|" /var/lib/asterisk/agi-bin/4747
Now you’re ready to try out the speech recognition apps. Dial 411 and say "American Airlines" to be connected to American.
To access Wolfram Alpha by phone, dial 4747 and enter your query, e.g. "What planes are overhead now?" Read the Nerd Vittles tutorial for additional examples and tips.
Configuring the Issabel Fax Server
Incredible PBX for Issabel includes turnkey fax support with Issabel. Once you have added a trunk that supports VoIP faxing (HINT: Google Voice trunks work great!), fax configuration with Issabel only takes a minute. Start by logging into the Issabel web interface as admin. First, navigate to PBX:PBX Configuration:Extensions:Fax and obtain your password for extension 329. Next, navigate to Fax:Virtual Fax:New Virtual Fax. Fill in the form as shown below using your actual email address and phone number for receiving faxes as well as your actual extension 329 secret. Then click SAVE. Assuming you typed your secret correctly, you will see a status notification showing virtual fax machine "Running and idle on ttyIAX1."
Assuming you already have set up a Google Voice trunk as outlined above, the next step is to modify the Inbound Route for this trunk to support fax detection. In that way, incoming fax calls will automatically be redirected to extension 329 and the received faxes will be emailed to you in PDF format. Set the email address in Fax:Fax Master. In addition, the faxes can be downloaded and managed from Fax:Virtual Fax:Fax Viewer. Modify your Inbound Route to match the #3 settings shown below. Then save/reload your changes.
To receive the incoming faxes by email, navigate to Fax:Fax Master and enter your email address. Then click SAVE.
The final step is to designate the IP addresses of those authorized to send faxes using Issabel. Navigate to Fax:Fax Clients and specify the public and private IP addresses (one per line) authorized to send faxes. Then click SAVE. Hylafax clients can be used remotely, or you can use the web utility included with Issabel: Fax:Virtual Fax:Send Fax.
The best way to test things out is to send yourself a test fax. FaxZERO lets you send 5 free faxes of up to 3 pages every day. Give it a whirl.
To send a fax out from your server from the Linux CLI using either a text document or PDF file, the syntax looks like the following:
sendfax -n -d 8005551212 smsmsg.txt
Sampling Other Incredible PBX Applications
As installed, Incredible PBX includes dozens of additional applications for Asterisk. Here’s how to sample some of them using a softphone connected to your Issabel PBX. A good place to start is Allison’s Demo IVR (dial D-E-M-O) using any phone connected to your PBX:
Nerd Vittles Demo IVR Options
1 – 411 -Call by Name (say "American Airlines")
2 – 2663 – MeetMe Conference
3 – 4747 – Wolfram Alpha
4 – 53669 – Lenny (The Telemarketer’s Worst Nightmare)
5 – 951 – Today’s News Headlines
6 – 947 – Weather Forecast (enter a 5-digit ZIP code)
7 – 86329 – Today in History
8 – 701 – Speak to a Real Person
For ODBC demos, dial 222 and enter 12345 for the employee number for a sample database application. Or dial 223 for a sample ODBC dialer using AsteriDex. Enter 263 (first three letters of American Airlines) to place the call. Sample dialplan code is stored in /etc/asterisk/odbc.conf. Dial L-E-N-N-Y (53669) to call or forward telemarketer calls to Lenny. Dial T-I-M-E (8463) for Time of Day. Dial *88HHMM to set an Alarm for HH:MM where HH is the hour of the day in military time. Dial C-O-N-F (2663) for MeetMe conference. Conference credentials are in /root/passwords.FAQ. Voice Dialer (411) works with any database entry in AsteriDex. Access AsteriDex with a browser at https://Issabel-IP-Address/asteridex4. Telephone Reminders can be scheduled by phone (123) or via the web: https://Issabel-IP-Address/reminders. Sample code for the FLITE, GoogleTTS, and PicoTTS engines is in 951 (Yahoo News) context of /etc/asterisk/extensions_custom.conf. All of your FreePBX "old favorites" including blacklists, call transfers and forwarding, dictation, recordings and more are still available as well: PBX:PBX Config:Feature Codes.
Update: We’ve added Allison’s Demo IVR to our own Issabel server at Vultr ($2.50/mo.)1 so you can judge the call quality and feature set for yourself. You can even send us a fax or SMS message if you’d like to try out those features:
For VoIP callers, use this free SIP URI: 1015954772235642@tampa.voip.ms
Published: Monday, July 10, 2017 Updated: Tuesday, July 25, 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…
- Some of our links refer users to providers that support Nerd Vittles through referral fees or advertising. These funds help cover the costs of our blog. We never recommend particular products solely to generate revenue. However, when pricing is comparable or particular features warrant our recommendation, we support these vendors and deeply appreciate their financial support of our software development efforts. [↩]
Independence Day for Elastix 4: Introducing Issabel 4
There was more than a little disappointment when PaloSanto Solutions closed up their Elastix® shop last year. The good news was that 3CX stepped in with a terrific free offer on their commercial PBX for Elastix users. And, of course, Sangoma® quickly learned Spanish to tout their closed source crippleware, the FreePBX® Distro. But today’s a new day, and there’s another great surprise as the discontinued Elastix 4 open source product is reincarnated as Issabel 4. Featuring a respin of CentOS 7 plus Asterisk® 11 and FreePBX GPL components (only), Issabel is the first new open source project in the Asterisk community in many, many years. And what a terrific GPL platform it is.
If you liked Elastix 4, you’re going to love Issabel. Its feature set not only matches Elastix 4, but it further enhances it with a collection of bells and whistles you won’t find anywhere else for free. In addition to a modern GUI and Dashboard, you’ll find a rich collection of add-ons to further enhance the FreePBX GPL components without any NagWare. There’s a native email server, a firewall, a fax server, chat service, Festival TTS, and optional add-ons for a Call Center, FOP 2, Vtiger CRM, and RoomX for hotel management.
2018 NEWS FLASH: Incredible PBX® 13 for the 2018 edition of Issabel supporting Asterisk 13 is now available for download. Complete tutorial available at this link.
For today, let’s get Issabel up and running. Installation is a breeze. Just download the 64-bit ISO from SourceForge. Be sure to use the January 2018 release. We will upgrade it from there when we install Incredible PBX 13 for Issabel in this new tutorial. Then burn the image to a CD or, better yet, load it up on your desktop PC with VirtualBox and take it for a quick spin. We think you’ll be thrilled.
Once you’ve installed the image and created your passwords, simply log into the Issabel GUI as admin with your GUI password. Click on PBX in the left column of the Dashboard. Set up a SIP trunk with an inbound and outbound route. Add an extension and a softphone, and you’ll be making your first call in minutes. If you’re unfamiliar with Elastix and FreePBX GPL components, here’s a quick tutorial to get you started.
Getting Started with Issabel 4
Here’s a quick overview of the installation and setup process for Issabel 4:
- Choose HW Platform – Dedicated PC, Cloud, Virtual Machine
- Install Issabel 4 – 64-bit CentOS 7 platform
- Activate Trunks for Issabel 4
- Connect a Softphone to Issabel 4
- Configuring SMTP Mail with Issabel 4
1. Choose a Platform for Issabel 4
Issabel 4 works equally well on dedicated hardware, a cloud-based server, or a virtual machine. Just be sure you have a sufficiently robust Internet connection to support 100Kb of download and upload bandwidth for each simultaneous call you wish to handle with your new PBX.
For Dedicated Hardware, we recommend at least an Atom-based PC of recent vintage with at least a 30GB drive and 4GB of RAM. That will take care of an office with 10-20 extensions and a half dozen or more simultaneous calls if you have the Internet bandwidth to support it. Our favorite hardware platform remains the $200 Intel NUC, and you can read all about it here.
For Cloud-Based Servers, we recommend RentPBX, one of our financial supporters who also happens to size servers properly and restrict usage solely to VoIP. This avoids performance bottlenecks that cause problems with VoIP calls. Yes, we have a coupon code for you to get the $15/month rate: NOGOTCHAS.1 The new image to support Issabel 4 should be available shortly. In the meantime, you can roll your own VPS using the Issabel ISO at Vultr for just $5 $2.50 a month using our referral code. First, upload the ISO by choosing Servers -> ISO -> Add ISO after login. ISO image link: http://nerd.bz/2ukpAXC
Next, create a new Instance by clicking the + icon in Servers. Choose a Server Location close to you. For Server Type, choose Upload ISO and pick your new ISO image. Use $2.50/month Server Size. Enter issabel for your Hostname. Leave everything else at default settings. Click Deploy Now. Click the Manage tab once your Cloud Instance is up and running. Make note of your server’s IP address. You’ll need it shortly.
Wait about a minute and then click View the Console to connect to VNC with your booted Issabel installer. Expand the console window so you can see the entire screen. Highlight Install and press ENTER. Choose the default Installation Destination, and then click Begin Installation. Click Root Password and set a very secure root password. Click Done to proceed with install. Have a cup of coffee while you wait. At about the 60% mark in the post-installation setup, the install may appear to be hung. Relax. It’s not. While you’re waiting, on the Vultr console, go to Settings -> Custom ISO under Server Details for your instance and locate the button to remove the ISO from the instance. Don’t do it yet. Switch back to the console window and wait for the instance to reboot and redisplay the Issabel installer menu. Now switch back to the Vultr window in your browser and click the Remove ISO button, and your server will reboot again. Close the Console window and then reopen it. Enter passw0rd (with a zero) for your MySQL password when prompted and create a very secure admin password for the web interface. Once the installer finishes, you can login to the IP address of your instance using user account admin and the web password you just created. If you ever need to reset them, login with SSH as root using your root password and issue the following command: issabel-admin-passwords --change
Update: On other cloud platforms that do not directly support ISO image installs, you can begin by installing a minimal, 64-bit version of CentOS 7. Then issue the following commands via SSH or Putty after logging into your server as root. NOTE: The 2018 release of Issabel featuring Asterisk 13 is NOT yet supported:
yum -y install wget nano openssh-server wget -O - http://repo.issabel.org/issabel4-netinstall.sh | bash
For Virtual Machine Installs, we recommend Oracle’s VirtualBox platform which runs atop almost any operating system including Windows, Macs, Linux, and Solaris. Here’s a link to our original VirtualBox tutorial to get you started. We suggest allocating 1GB of RAM and at least a 20GB disk image to your virtual machine for best performance. We actually used VirtualBox to build our first Issabel 4 platform.
2. Install 64-bit Issabel 4 on Your Platform
Begin by downloading the 64-bit Issabel 4 ISO. For dedicated hardware, burn the ISO image to a DVD and boot your server with the Issabel 4 ISO to begin the install. You’ll be presented with the CentOS 7 Installation GUI:
Choose: Time Zone (click)
Choose: Keyboard (click)
Choose: Install Drive (double-click)
Choose: Root Password (Make it Secure!)
Wait for Install and Reboot to Complete
Set MySQL Password to: passw0rd (MANDATORY: with a zero!)
Set Issabel admin Password: minimum 10 alphanumeric characters with upper & lowercase
For VirtualBox, create an Issabel 4 virtual machine of Linux (RedHat 64-bit) type by clicking New. Click Settings button. In System, enable I/O APIC and disable Hardware Clock in UTC Time. In Audio, enable Audio for your sound card. In Network, enable Bridged Adapter for Adapter 1. In Storage, click on Empty in the Storage Tree. Then click on the Disk icon to the right of CD/DVD Drive attributes. Choose the Issabel 4 ISO file that you downloaded. Click OK. Then start the virtual machine to begin the installation process. Follow the setup steps above to install Issabel 4 in your virtual machine.
3. Activate Trunks with Issabel 4
For those migrating from another aggregation including PIAF 3, this should be familiar territory for you. Using a browser, log into Issabel 4 at the IP address of your server using your admin password. Before you can actually make or receive calls outside your PBX, you’ll need at least one trunk. In the Issabel 4 GUI, click PBX -> PBX Configuration -> Trunks. Once you have your credentials from a provider, choose a provider from the list of preconfigured trunks on the right or create a new one. If you’re using one of the preconfigured options, remember to enable the trunk after adding your desired CallerID and credentials. Then save your settings and reload your Asterisk dialplan. That’s it. You’re ready to go.
To display your trunk registrations: asterisk -rx "sip show registry"
Google Voice Setup. If you wish to use Google Voice for free calling in the U.S. and Canada, you’ll need to sign up for an account with Simonics SIP to Google Voice Gateway service. Complete documentation is here. With the latest version of Incredible PBX 13 for Issabel, Google Voice support is included in the GUI.
Connection Issues. If you experience problems getting trunks to register with providers, add or remove the following entry in /etc/asterisk/sip_custom.conf: nat=yes. Then restart Asterisk: amportal restart.
4. Configure a Softphone for Issabel 4
You can connect virtually any kind of telephone to your Issabel 4 PBX after you first create a SIP extension. Plain Old Phones require an analog telephone adapter (ATA) which can be a separate board in your computer from a company such as Digium. Or it can be a standalone SIP device such as ObiHai’s OBi100 or OBi110 (if you have a phone line from Ma Bell to hook up as well). SIP phones can be connected directly so long as they have an IP address. These could be hardware devices or software devices such as the YateClient softphone. We’ll start with a free one today so you can begin making calls. You can find dozens of recommendations for hardware-based SIP phones both on Nerd Vittles and the PIAF Forum when you’re ready to get serious about VoIP.
We recommend YateClient which is free. Download it from here. Run YateClient once you’ve installed it and enter your credentials once you create a 701 extension in Issabel 4. You’ll need the IP address of your server plus your extension 701 password. Fill in the blanks using the IP address of your server, 701 for your account name, and whatever password is assigned to the extension. Here’s what your entries should look like. Click OK to save your entries.
Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place some test calls.
5. Configuring SMTP Mail with Issabel 4
Outbound email support using Postfix is preconfigured with Issabel 4. You can test whether it’s actually working by issuing the following command using your destination email address after logging in as root:
echo "test" | mail -s testmessage yourname@gmail.com
If you don’t receive the email message within a minute or two and you’ve checked your spam folder, chances are your ISP is blocking downstream SMTP servers in an effort to combat spam. Comcast is one of the usual suspects. To enable outbound email service for delivery of voicemail and other email messages with a provider blocking downstream SMTP servers, just follow our recent tutorial to reconfigure Postfix on your server.
Published: Monday, July 3, 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…
- Some of our links refer users to providers that support Nerd Vittles through referral fees or advertising. These funds help cover the costs of our blog. We never recommend particular products solely to generate revenue. However, when pricing is comparable or particular features warrant our recommendation, we support these vendors and deeply appreciate their financial support of our software development efforts. [↩]
The World Traveler and 3CX: A Match Made in Heaven
Last week we introduced the new (free) version of PIAF5 powered by 3CX v15.5 supporting four simultaneous calls, unlimited trunks, 10 extensions, and 10-user conference calls. And today we’re torture-testing our new 3CX server in the Bahamas aboard one of Carnival’s 3,000-passenger floating cities. Somebody’s gotta do it, right? What makes this such a challenging test for any PBX are several things. First, we’re using a free Google Voice trunk on a free 3CX PBX that we configured in under 10 minutes at CloudAtCost for a one-time cloud server charge of $17.50. Second, we’re sharing a satellite Internet connection with 3,000 other people in the middle of the Caribbean. The weekly charge is about $100 so every Internet junkie subscribes. Third, we’re using a 3CX Client on an iPhone in Airplane Mode. And, finally, we’re sitting behind the most Draconian firewall you can imagine because Carnival assumes everyone is a bad guy trying to bring their Internet service to its knees.
For those coming from the Asterisk® world, I don’t have to remind you how challenging this NAT-based setup would be even assuming you had a flawless Internet connection. Believe me. We don’t. And the secret sauce that makes all of this seem like child’s play is the latest collection of 3CX Clients for PCs, Macs, Android devices, and iPhones/iPads. Simply download the client for your platform, log into your 3CX portal and send the welcome email from a configured extension to your phone, open the email on your phone and double-click on the attachment, and boom. Your 3CX Client is automatically configured in seconds and ready to make your first call. A monkey could do it. It’s that easy!
So our torture-test for today looks more like a final exam in VoIP telephony. We’ll be using Carnival’s WiFi connection from our iPhone with its iOS 3CX Client. We’ll dial into the Incredible PBX™ at our office in Charleston. The office number is configured with a Stealth AutoAttendant which we’ll use to make an outbound call to our Demo IVR in Marbella, Spain using DISA and a FreeVoipDeal trunk. For the techies, it’s the NAT Trifecta with DTMF hurdles that are virtually impossible to traverse using Asterisk and any SIP client.
Guess what? It not only works, but it sounds like you’re sitting in the adjoining office. No echo, no DTMF problems, no missing audio, and no detectable problems in voice quality with either the Charleston IVR or the Marbella IVR. If cost matters and traveling is a key component in your telephony requirements, you owe it to yourself to set up a free 3CX PBX and take it for a spin. Whether you use it to supplement an existing Asterisk setup or as a standalone PBX, we think you’ll be thrilled with the results.
Continue reading about the new, free PIAF5 server powered by 3CX v.15.5…
Originally published: Monday, June 12, 2017
Need help with 3CX or VoIP? 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…
Free Asterisk Solutions