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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiRTPbleed Security Alert: Asterisk Calls Can Be Intercepted
If you’ve installed Asterisk® during the past 4½ years, your server has a MAJOR security problem. If you didn’t already know, with Asterisk, your VoIP conversations actually are carried over a random UDP port using the Real Time Protocol (RTP), not the SIP port (UDP 5060) which handles the setup and teardown of your VoIP connections. It turns out that, since March 2013, all of that RTP traffic and thus your conversations could be intercepted and redirected by anyone on the Internet. As this recent article in The Register noted:
The problem occurs when [communications] systems like IP telephony have to get past network address translation (NAT) firewalls. The traffic has to find its way from the firewall’s public IP address to the internal address of the device or server, and to do that, RTP learns the IP and port addresses to associate with a call.
The problem is, the process doesn’t use any kind of authentication.
This is exacerbated by the fact that, by default, Asterisk and FreePBX® traditionally use the NAT=yes setting (whether needed or not) to enable this navigational magic just in case your calls need it. Without it, you may end up with no audio or one-way audio on your calls. Traditional wisdom was that an attacker needed to be positioned between the caller and the Asterisk server in order to intercept this media stream. As luck would have it, it turns out the man in the middle didn’t need to be in the middle after all. He could be anywhere on the Internet. The old adage to talk on the phone as if someone else were listening turns out to have been pretty good advice in the case of Asterisk communications. Even if you had a firewall, chances are you protected UDP port 5060 while exposing and forwarding UDP 10000-20000 to Asterisk without any safeguards.
According to last week’s Asterisk advisory, “To exploit this issue, an attacker needs to send RTP packets to the Asterisk server on one of the ports allocated to receive RTP. When the target is vulnerable, the RTP proxy responds back to the attacker with RTP packets relayed from the other party. The payload of the RTP packets can then be decoded into audio.” Specifically, if UDP ports 10000-20000 are publicly exposed to the Internet, anybody and everybody can intercept your communications without credentials of any kind. WOW!
So, there’s a patch to fix this, right? Well, not exactly:
Note that as for the time of writing, the official Asterisk fix is vulnerable to a race condition. An attacker may continuously spray an Asterisk server with RTP packets. This allows the attacker to send RTP within those first few packets and still exploit this vulnerability.
The other recommended "solutions" aren’t much better:
- When possible the nat=yes option should be avoided
- To protect against RTP injection, encrypt media streams with SRTP
- Add config option for SIP peers to prioritize RTP packets
The nat=no option doesn’t work if you or your provider employs NAT-based routers. The SRTP option only works on more recent releases of Asterisk, and it also requires SRTP support on every SIP phone. Prioritizing RTP packets is not a task for mere mortals.
Surprisingly, the one solution that is not even mentioned is hardening your firewall to block incoming UDP 10000-20000 traffic that originates outside your server. Our recognized SIP expert on the PIAF Forum had the simple solution. Bill Simon observed:
If the SDP in the INVITE or subsequent re-INVITE contains routable IP addresses, then use them for media. If the SDP contains non-routable IP addresses, then the client is behind a NAT and not using any NAT traversal techniques like SIP ALG, ICE/STUN, so send to the originating IP. Why are we making allowances here for media to come from anywhere? I think you can probably clamp down your firewall as much as you want, because symmetric RTP should allow media to get through by way of establishing an outbound stream (inbound stream comes back on the same path).
Our testing confirms that simply blocking incoming RTP traffic on your firewall solves the problem without any Asterisk patch. In short, RTP traffic cannot originate from anonymous sources on the Internet.
For those using Incredible PBX® or Travelin’ Man 3 or an IPtables firewall, the fix is easy. Simply remove or comment out the INPUT rule that looks like this and restart IPtables:
-A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
On RedHat/CentOS servers, the rule is in /etc/sysconfig/iptables. On Debian/Ubuntu and Raspbian servers, you’ll find the rule in /etc/iptables/rules.v4. On Incredible PBX for Issabel servers, you’ll find the rule in /usr/local/sbin/iptables-custom. On all Incredible PBX platforms, remember to restart IPtables using only this command: iptables-restart.
Published: Friday, September 8, 2017
Need help with Asterisk? Visit the PBX in a Flash Forum.
Special Thanks to Our Generous Sponsors
FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.
BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.
The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.
VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
Some Recent Nerd Vittles Articles of Interest…
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.
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…
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. [↩]
Meet the New Incredible Fax: A $10 Fax Machine for Asterisk
Every year, technology gets better and cheaper. And, believe it or not, that even includes faxing especially with Asterisk® and Incredible PBX™. Today we take another giant leap forward by introducing fax technology with the $10 Raspberry Pi Zero W. Adding a free Google Voice trunk provides not only free calling in the U.S. and Canada, but now you also get free faxing as well. And the latest release of Incredible PBX lets your RasPi determine whether incoming calls are humans or faxes, and it’ll route them accordingly. To send faxes, you can use the bundled AvantFax GUI client which lets you send PDF documents as faxes with a couple button clicks.
Overview. Once you’ve downloaded the Incredible PBX for RasPi image and made yourself a microSD card for your RasPi, the setup goes like this. First, we’ll configure a WiFi connection to support your server. Then we’ll install a Google Voice trunk. Next, we’ll use the included Incredible Fax installer to put the HylaFax and AvantFax components in place and to set up an email address for delivery of incoming faxes in PDF format. And finally we’ll use the GUI to configure your Google Voice trunk to manage incoming calls from both fax machines and individuals that actually want to talk to you. Faxes will be delivered to your email address, and traditional calls will be routed to a SIP phone or smartphone of your choice. In under 30 minutes, you’ll have a plug-and-play computer that’s about the size of a couple sticks of chewing gum.
Raspberry Pi Zero W Shopping List
Before you can install Incredible PBX, you’ll need a compatible Raspberry Pi Zero W platform. Unless you already have some of the components, the easiest way to begin is to purchase a bundle that includes all the components you’ll need. Here’s your best bet. It’s $35 and includes everything except a USB keyboard and an HDMI monitor and cable. Click on the image for ordering info:
Incredible PBX Installation Tutorial
Here’s everything you need to know about installation and setup. Just follow the links.
- Download and unzip Incredible PBX image from SourceForge (with GV OAuth support!)
- Transfer Incredible PBX image to microSD card
- Boot Raspberry Pi Zero W from new microSD card
- Login to RasPi console as pi:raspberry to initialize your server and configure WiFi
- Reboot after writing down your server IP address
- Login via SSH as root:password to secure your passwords & configure firewall
- Download latest Incredible Fax installer: http://nerd.bz/2nSeHKs
- Install Incredible Fax: /root/incrediblefax13_raspi3.sh (Credentials: admin:password)
Once everything is set up and working, you can remove the keyboard and monitor and put the Raspberry Pi Zero W on a shelf and run it with nothing more than a power adapter. Each time you reboot Incredible PBX, you’ll get an email with the IP address of your server. The recommended setup is to reserve the IP address assigned by your DHCP server in your router’s configuration. Then you have the equivalent of a static IP address while preserving the flexibility to move your RasPi to another network if the need ever presents itself.
Incredible PBX Initialization Steps
With a USB keyboard and HDMI monitor attached to your RasPi, power up the device. Login as pi with the password: raspberry. Incredible PBX first will whir through a few initialization steps. As part of the Raspbian OS, the Raspberry Pi Foundation includes a handy utility called raspi-config. This gets run automatically as part of the initial setup procedure in Step #4. At a minimum, you should configure the following options:
- Expand Filesystem (to use your entire microSD card)
- Wait for Network at Boot (choose Slow to Enable)
- International Options (configure all four options)
- Advanced Options – HostName (name your server)
- Finish (Save your settings but Delay reboot)
Incredible PBX WiFi Setup
The Raspberry Pi Zero W has WiFi-only networking. To get Internet connectivity, you’ll need to configure your server in Step #4 so that it can find your WiFi Access Point. Edit /etc/wpa_supplicant/wpa_supplicant.conf and insert the SSID and password (psk) for your WiFi network. Then save the file. Finally, stop and restart the wlan0 interface on your RasPi device, count to 15, and check the status of your server to decipher the new IP address for your WiFi connection. If no WiFi address, rinse and repeat.
ifdown wlan0 ifup wlan0 pbxstatus
Incredible PBX Phase 2 Configuration
Once you have your network IP address in hand, reboot your RasPi: reboot. Then use SSH or Putty to login to your RasPi from your desktop computer. The credentials are root:password. Complete the setup process by answering the prompts and be sure to set up a very secure root password and GUI admin password for your server.
Configuring a Gmail SmartHost for SendMail
Because of spammers, most Internet service providers now block mail from downstream mail servers. Your RasPi qualifies. There’s an easy way to determine if email service from your server is blocked. Just run the following command substituting your email address. Be sure to check your inbox and spam folder to determine whether you received the email.
echo "This is a test message." | mail -s testmessage yourname@youremaildomain.com
If you flunked the test, here’s what to do next. Modify SendMail to use an existing Gmail account as a SmartHost for email delivery. This means Gmail will actually send the messages rather than your server. Log into your RasPi as root and issue these commands:
cd /etc/mail hostname -f > genericsdomain touch genericstable makemap -r hash genericstable.db < genericstable mv sendmail.mc sendmail.mc.original wget http://incrediblepbx.com/sendmail.mc.gmail cp sendmail.mc.gmail sendmail.mc mkdir -p auth chmod 700 auth cd auth echo AuthInfo:smtp.gmail.com \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" > client-info echo AuthInfo:smtp.gmail.com:587 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info echo AuthInfo:smtp.gmail.com:465 \\"U:smmsp\\" \\"I:user_id\\" \\"P:password\\" \\"M:PLAIN\\" >> client-info nano -w client-info
When the nano editor opens the client-info file, change the 3 user_id entries to your Gmail account name without @gmail.com and change the 3 password entries to your actual Gmail password. Save the file: Ctrl-X, Y, then ENTER.
Now issue the following commands. In the last step, press ENTER to accept all of the default prompts:
chmod 600 client-info makemap -r hash client-info.db < client-info cd .. make sed -i 's|sendmail-cf|sendmail\/cf' /etc/mail/sendmail.mc sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/Makefile sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/databases sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.mc.gmail sed -i 's|sendmail-cf|sendmail\/cf|' /etc/mail/sendmail.cf.errors sendmailconfig
Finally, stop and restart SendMail and then send yourself a test message. Be sure to check your spam folder!
/etc/init.d/sendmail stop /etc/init.d/sendmail start apt-get install mailutils -y echo "test" | mail -s testmessage yourname@yourdomain.com
Check mail success with: tail /var/log/mail.log. If you have trouble getting a successful Gmail registration (especially if you have previously used this Google account from a different IP address), try this Google Voice Reset Procedure. It usually fixes connectivity problems. If it still doesn’t work, enable Less Secure Apps using this Google tool.
The last step is to add the following command to /etc/rc.local to send you an email with your IP address and SSID whenever the RasPi is rebooted. Insert the following commands just above the exit 0 line at the end of the file. Use an email address to which you have access on the road!
echo "IP address for your Raspberry Pi: $(hostname -I) plus wireless network, if any: `iwconfig`" | mail -s "Raspberry Pi IP Address" yourname@yourdomain.com
Installing a Google Voice Trunk for Free Calling
If you want to use Google Voice, you'll need a dedicated Google Voice account to support Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!
IMPORTANT: Do NOT under any circumstances take Google’s bait to switch from Google Chat to Hangouts, or you may forever lose the ability to use Google Chat with Incredible PBX. Also be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That's the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. Good News! You're in luck. Google has apparently had a change of heart on discontinuing Google Chat support so it's enabled by default in all new Google Voice accounts. Once you've created a Gmail and Google Voice account, go to Google Voice Settings and click on the Calls tab. Make sure your settings match these:
- Call Screening - OFF
- Call Presentation - OFF
- Caller ID (In) - Display Caller's Number
- Caller ID (Out) - Don't Change Anything
- Do Not Disturb - OFF
- Call Options (Enable Recording) - OFF
- Global Spam Filtering - ON
Click Save Changes once you've adjusted your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.
UPDATE: Google has improved things... again. You may not see the options documented above at all. Instead, you may be presented with the new Google Voice interface which does not include the Google Chat option. But fear not. At least for now there's still a way to get there. After you have set up your new phone number, click on (1) Settings -> Phone Numbers and then click (2) Transfer (as shown below). That returned the old UI. Make sure the Google Chat option is selected and disable forwarding calls to default phone number.
One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!
Now you're ready to configure your Google Voice account in Incredible PBX. First, you'll need to obtain an OAuth 2 token for your Google Voice account. For a complete Google Voice OAuth tutorial, follow steps 8-10 in this Nerd Vittles tutorial. Once you have your credentials, you can do the rest of the Google Voice setup from within the Incredible PBX GUI. Choose Connectivity -> Google Voice. Once you've entered your credentials, you MUST restart Asterisk from the command line, or Google Voice calls will fail: amportal restart
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.
Another option is to use an inexpensive SIP Gateway to Google Voice. The Simonics trunk in the Incredible PBX GUI is preconfigured for this purpose. All you'll need is your Google Voice credentials. Get started with this tutorial.
Installing Incredible Fax with HylaFax & AvantFax
Once you complete the initial configuration and get your mail server and Google Voice trunk squared away, it's time to run the Incredible Fax installer: ./incrediblefax13_raspi3.sh. You'll be prompted for an email address to which to deliver incoming faxes. After that, everything is pretty much automatic. A few prompts will appear during the installation process. Just press the ENTER key each time and ignore any errors you might see scrolling across your screen. They're harmless. When the HylaFax and AvantFax installs finish, reboot your server. Faxing won't work until you do!
Next, you need to change the default password for AvantFax which is a web-based interface to send faxes and read incoming faxes. From a browser, log into the IP address of your server. When the Incredible PBX menu appears, click the Users tab to display the Administrator menu. Then click on the AvantFax icon to load AvantFax. When prompted for credentials, enter admin:password for your username and password. You'll be prompted to change your password. Make it secure!
Finally, we need to configure your PBX to properly answer calls from fax machines as well as humans. Return to the Incredible PBX Admin menu and click the Incredible GUI icon. Then click the Server Administration icon. When prompted for your username and password, enter admin and the password you configured when you set up your server above. When the System Status screen displays, choose Connectivity -> Inbound Routes -> Default. Make the bottom section of the template look like this substituting your desired Destination for voice calls if you don't want them sent to the Incredible PBX IVR. Click Submit to save your changes and then reload your dialplan when prompted.
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. When you're ready to send a fax from Incredible PBX, log back into AvantFax, click on the Send Fax icon, and follow your nose.
Mastering the Incredible PBX Feature Set
Now would be a good time to explore the Incredible PBX applications. Continue reading there. If you have questions, join the PBX in a Flash Forums and take advantage of our awesome collection of gurus. There's an expert available on virtually any topic, and the price is right. As with Incredible PBX, it's absolutely free. Enjoy!
Originally published: Monday, March 27, 2017 Updated: Friday, May 12, 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...
IBM’s Speech Recognition Engine Comes to Asterisk
Eight years ago, we introduced transcription for Asterisk® voicemail messages. When the messages were delivered by email, you got both a recording and the transcribed text courtesy of Google. As with most things Google, the licensing terms changed regularly and voicemail transcription became more convoluted until it became next to worthless. Today we begin our new exploration of IBM’s Watson Developer Cloud. It offers a rich collection of services at unbelievably low price points. We’re kicking things off by introducing a better Speech-to-Text (STT) solution with IBM’s Bluemix. The STT API performs better than any speech recognition engine in the world. And you won’t have to worry about Google breaking our middleware every month. On the Lite plan, up to 100 minutes per month are free. Or you can opt for the Standard pay-as-you-go plan for 2¢ per minute and let your customers yack all they like. That works out to $1.20 an hour which still is pretty cheap secretarial help. In coming weeks, we will introduce IBM’s Text-to-Speech (TTS) offering and Lisa. Up to a million characters of TTS service monthly are free. Here’s a sample to give you a taste of the voice quality:
[soundcloud url="https://api.soundcloud.com/tracks/312693441″ params="auto_play=false&hide_related=false&show_comments=true&show_user=true&show_reposts=false&visual=true" width="100%" height="350″ iframe="true" /]
NOV. 1 UPDATE: IBM has moved the goal posts effective December 1, 2018:
For new deployments, your API Username will be apikey, and your API Password will be your actual APIkey.
Overview. What we’ve done today is integrate the STT Bluemix API directly into existing Asterisk voicemail systems. We started with Nicolas Bernaerts’ terrific sendmailmp3 script. It works on both the Wazo and FreePBX® platforms. If you have deployed Incredible PBX, then the setup takes a couple of minutes. For everyone else, there’s an additional configuration step using your favorite GUI. To get started, you’ll sign up for a Bluemix account and obtain your credentials. Next, you download today’s script for your platform and insert your credentials. Finally, you set up voicemail on the extensions desired and insert an email address for each voicemail account. On generic FreePBX systems, you’ll need to add the name of our script to manage your voicemail recordings.
What About the Quality? Here’s the bottom line. Speech recognition isn’t all that useful if it fails miserably in recognizing everyday speech. The good news is that IBM Watson’s speech recognition engine is now the best in the business. If you want more details, read the article below which will walk you through IBM’s latest speech recognition breakthrough:
Why IBM's speech recognition breakthrough matters for AI and IoT. Via @techrepublic https://t.co/AJi8MA3E20
— IBM Developer (@IBMDeveloper) March 15, 2017
Creating an IBM Bluemix Speech to Text Account
Follow this link to set up your IBM account and obtain credentials for both Speech to Text (STT) and Text to Speech (TTS) services. Please note that your STT and TTS API keys will NOT be the same. So don’t accidentally use the wrong one.
Installing STT Engine with Incredible PBX for Wazo
1. After logging into your Incredible PBX for Wazo server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm.tar.gz tar zxvf sendmailibm.tar.gz rm -f sendmailibm.tar.gz
2. Edit sendmailibm and insert Bluemix STT credentials on lines 29 and 30. Save the file.
3. Edit bluemix-test and insert Bluemix STT credentials on first two lines. Save the file.
4. Copy the updated sendmailibm file to sendmail:
cd /usr/sbin cp -p sendmailibm sendmail
5. Test your Bluemix STT setup: bluemix-test
6. Result should be: please record your message after the beep
7. Set up voicemail account for a Wazo extension with your email address.
8. Place a test call to the extension and record a voicemail when prompted.
9. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX for RasPi
1. After logging into your Raspberry Pi server as root using SSH/Putty:
cd /usr/sbin wget http://incrediblepbx.com/sendmailibm-raspi.tar.gz tar zxvf sendmailibm-raspi.tar.gz rm -f sendmailibm-raspi.tar.gz
2. Edit sendmailmp3.ibm and insert Bluemix STT credentials on lines 28 and 29. Save file.
3. Edit bluemix-test and insert Bluemix STT credentials on first two lines. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Test your Bluemix STT setup: bluemix-test
6. Result should be: your dictation is now being processed and emailed please wait
7. Set up voicemail for a RasPi extension with your email address.
8. Place a test call to the extension and record a voicemail when prompted.
9. Your message will be transcribed and delivered via email.
Installing STT Engine with Incredible PBX 13
1. After logging into your Incredible PBX 13 server as root using SSH/Putty:
cd /usr/local/sbin wget http://incrediblepbx.com/sendmailibm-13.tar.gz tar zxvf sendmailibm-13.tar.gz rm -f sendmailibm-13.tar.gz
2. Edit sendmailmp3.ibm and insert Bluemix STT credentials on lines 28 and 29. Save file.
3. Edit bluemix-test and insert Bluemix STT credentials on first two lines. Save the file.
4. Copy the updated sendmailmp3.ibm file to sendmailmp3:
cd /usr/local/sbin cp -p sendmailmp3.ibm sendmailmp3
5. Test your Bluemix STT setup: bluemix-test
6. Result should be: we are now transferring you out of the company directory…
7. Set up voicemail for an extension and include your email address.
8. Place a test call to the extension and record a voicemail when prompted.
9. Your message will be transcribed and delivered via email.
Installing STT Engine with Legacy FreePBX® Servers
1. Follow steps #1 through #7 from the Incredible PBX 13 tutorial above.
2. Choose Settings -> Voicemail Admin -> Settings in the GUI.
3. In the format field, insert: wav|wav49
4. In the mailcmd field, insert: /usr/local/sbin/sendmailmp3
5. Click Submit to save your settings and then Reload the FreePBX Dialplan.
6. Place a test call to the extension and record a voicemail when prompted.
7. Your message will be transcribed and delivered via email.
Update: Matt Darnell reports that, depending upon your existing setup, you may need to add the unix2dos and lame packages with legacy FreePBX servers to get MP3 messages delivered correctly.
Originally published: Monday, March 20, 2017
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