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The Most Versatile VoIP Provider: FREE PORTING
JUST RELEASED: Visit the Incredible PBX WikiBest of Both Worlds: Marrying Asterisk to 3CX’s Free PBX with a $35 Raspberry Pi
One of the real beauties of Asterisk® has always been its flexibility in talking to other PBXs, both commercial and open source. There are numerous reasons why you might want to try this. First, it makes it easy to migrate to a commercial platform where you can get support for mission critical telephony requirements. Second, you may want a hybrid setup where servers with on-site support personnel can run Asterisk while remote satellite offices can take advantage of a commercial PBX and the support options it offers. Third, you may want to take advantage of specific features that are only available by relying upon multiple PBX solutions. In the case of 3CX, their integrated softphone clients with one-click setup simplicity, conferencing and WebRTC apps, and Call Center offerings are the best in the business while providing unmatched VoIP security. Asterisk on the other hand is light-years ahead of almost everybody in the text-to-speech and voice recognition fields while offering the most powerful VoIP toolkit to build any custom VoIP application imaginable.
Today we thought it would be fun to walk you through the easy way to tie an Incredible PBX server with all its features to a powerful (free) 3CX platform with its virtually flawless softphone clients.1 When we’re finished, you’ll have a free 3CX server in the Cloud at a one-time total cost of $17.50. And you’ll be able to place and receive free U.S./Canada calls from any iPhone, Android phone, or PC using the 3CX client from anywhere in the world with nothing more than a WiFi connection. The Google Voice trunk supporting the calls will reside on Incredible PBX for the Raspberry Pi. When you’re sold on the power of the 3CX platform, you can upgrade to the 3CX 4-simultaneous call commercial offering with unlimited users and trunks at an annual cost of just $149. Maintenance and upgrades are included. Large organizations have relied upon back office servers for custom applications forever. And now you can take advantage of the same flexibility using a tiny $35 Raspberry Pi and our free (as in really free) Incredible PBX software. No Gotchas!
Initial Raspberry Pi Platform Setup
Before we can interconnect 3CX’s Free PBX with a Raspberry Pi, you obviously have to set up both PBX platforms. For the Raspberry Pi, our recent Nerd Vittles tutorial will walk you through the setup process. In lieu of a Raspberry Pi, you can use any legacy FreePBX®-based Asterisk platform including Incredible PBX 13, PIAF3, Elastix®, AsteriskNOW®, or FreePBX Distro®. The setup procedure is exactly the same.
Building a 3CX Server in the Cloud
Building a 3CX server in the Cloud is equally easy. Let’s go through the process once again. If you’re just experimenting, a lifetime Cloud-based server at CloudAtCost for a one-time charge of $17.50 cannot be beat. We would hasten to add that we don’t recommend this platform for production use, but it’s a terrific proof-of-concept option. When you’re actually ready to deploy 3CX for production use, the least costly Cloud solution is the $3.49 per month OVH RAID offering with 2GB of RAM and 10GB storage. The $5 per month offerings from Digital Ocean and Vultr are other alternatives worth a look. Both of these platforms come with free credits ($10 and $20, respectively) to let you try things out.
To get started, sign up for a $17.50 server at Cloud at Cost. They will send you credentials to log into the Cloud at Cost Management Portal. Change your password IMMEDIATELY after logging in. Just go to SETTINGS and follow your nose.
To build your free 3CX PBX, create a virtual machine by clicking on the CLOUDPRO button in the CloudAtCost control panel. Then click Add New Server. Choose 1 CPU, 512MB RAM, and 10GB storage for your server. Choose Debian 8 64bit as the OS Type and click Complete.
While CloudAtCost is building your server platform, obtain a free license key for 3CX.
Once the Debian 8 server appears in your Control Panel, it will look something like what’s shown above, not CentOS obviously. The red arrow points to the i button you’ll need to click to decipher the password for your new virtual machine. You’ll need both the IP address and the password for your new virtual machine in order to log into the server which is now up and running with a barebones Debian 8 operating system. Note the yellow caution flag. That’s telling you that Cloud at Cost will automatically shut down your server in a week to save (them) computing resources. You can change the setting to keep your server running 24/7. Click Modify, Change Run Mode, and select Normal – Leave Powered On. Click Continue and OK to save your new settings.
Finally, you’ll want to change the Host Name for your server to something more descriptive than c7…cloudpro.92… Click the Modify button again and click Rename Server to make the change. Your management portal then will show the new server name as shown above.
Next, log in to your new Debian server as root using SSH or Putty and issue the commands below. Step #1 is to change your root password. What appears as the fourth line below is actually part of the third line and needs to be run as a single command. The last line to install SendMail will actually be run after you elect to use the Web Interface Wizard to configure 3CX. Just run it from the SSH command line before you switch to a browser to complete the 3CX setup.
passwd wget -O- http://downloads.3cx.com/downloads/3cxpbx/public.key | apt-key add - echo "deb http://downloads.3cx.com/downloads/3cxpbx/ /" | tee /etc/apt/sources.list.d/3cxpbx.list apt-get update rm -f /zang-debian.sh apt-get -y install 3cxpbx apt-get -y install sendmail sendmail-bin
When the initial setup finishes, choose the Web Interface Wizard and complete the install using your favorite web browser. Enter your 3CX license key when prompted. Make up a very secure Username and Password to access your 3CX portal. Specify that your IP address is Dynamic when prompted (even though it isn’t). This tells 3CX to generate an FQDN for your server. Accept the default ports for HTTP (5000) and HTTPS (5001) access to your server. We recommend choosing 4-digit extensions numbers which will make it easy to distinguish 3CX extension numbers from 3-digit extension numbers of the RasPi platform. While logged into the 3CX management portal, adjust Settings → Email to Mail Server → 127.0.0.1 and Reply to → noreply@YourActual3CX-FQDN. Leave the other settings blank and click TEST then OK. Now download your favorite 3CX smartphone client, send yourself the Welcome Email for your default extension, and your 3CX initial setup is complete.
Server Interconnection Overview
Now we’re ready to interconnect the two servers. What we’ll be doing is creating Trunks on both the Raspberry Pi and the 3CX server and tying them together. We’ll use this trunk to handle the call traffic between the two PBXs. Then we’ll add incoming and outgoing call routes on both servers to specify how the individual calls should be routed. Because the free version of 3CX limits the administrator to a single trunk, we’ll offload all of the provider trunks to the Raspberry Pi and reserve the one available 3CX trunk as the interconnect path to the Raspberry Pi. For today’s setup, we’ll use 3CX’s free softphone clients as the actual phone devices for end-users. Of course, you could also use your favorite SIP phones, and 3CX provides automatic configuration for dozens of devices. But we want to introduce the 3CX smartphone clients because they provide an incredibly easy way to get users connected without having to worry about punching holes in firewalls.
To place outbound calls on the 3CX side, 3CX provides enormous flexibility in call routing. Because we chose 4-digit local extensions when we set up the 3CX server, it will make it easy to route other calls through the outbound trunk to the Raspberry Pi using nothing more than the length of the dial string. For example, 3-digit calls line up perfectly with extension numbers on the Incredible PBX for RasPi platform. So 3CX users can easily reach extensions connected directly to the Raspberry Pi. And 10-digit 3CX calls will be forwarded to the Raspberry Pi as traditional outbound calls. They will be processed just as if you had dialed a 10-digit call from a Raspberry Pi extension. For example, if you have a registered Google Voice trunk to handle 10-digit calls on the Raspberry Pi, then the same call path would be used for calls originating from 3CX extensions. And, yes, calls to the U.S. and Canada would still be free and would display the CallerID associated with the Raspberry Pi’s Google Voice trunk. You could get more creative and add an additional dialing prefix on the 3CX side to route specific types of calls to a designated outbound trunk on the Raspberry Pi side based upon the dialing prefix, but we’ll leave that as a homework project for you.
For incoming calls on the 3CX side, in addition to 4-digit local extension-to-extension calling, we can define the destination for incoming calls that originate from either a Raspberry Pi extension or from outside calls coming in from one of the Raspberry Pi’s provider trunks. These are managed by assigning one or more DIDs in the 3CX trunk configuration and then creating 3CX Inbound DID Rules that tell 3CX where to route calls to each defined DID. For 3CX softphone clients registered to extensions, it means your cellphone will ring whenever a call is routed to that particular extension. On the Raspberry Pi side, we create Incoming Call Routes for each DID to be routed to 3CX and specify our defined 3CX trunk as the destination for incoming calls from those DIDs. Not all DIDs on the Raspberry Pi have to be routed to the 3CX server obviously. That is merely one of many call destination options available to the administrator on the Raspberry Pi server.
Here’s a typical call path for an outside call that is placed to a Google Voice number registered with your Raspberry Pi. The Asterisk server running on the Raspberry Pi would answer the call placed to the Google Voice Trunk. Asterisk then would check for an Incoming Route on the Raspberry Pi with a DID matching the number of your Google Voice trunk. Finding a match, Asterisk would check for the desired destination of the call and would note that it is listed as the registered 3CX trunk. Asterisk would pass the call through this trunk to the 3CX server including its associated DID and CallerID info. The 3CX server would answer the incoming call and would check for an Incoming Route matching the DID passed from Asterisk. Finding a match, it would pass the call to the Extension specified in the Incoming Route. When 3CX rings the extension, it would also detect that a softphone was registered to that extension and would also ring the 3CX client on the user’s smartphone. The user answers the call on the 3CX client of their smartphone and begins a conversation. The free version of the 3CX server supports 8 simultaneous calls so you are unlikely to ever run out of call paths for calls in the home and small office environment.
Firewall Setup for Server Interconnection
Because the 3CX server is sitting in the Cloud, its firewall is configured automatically as part of the setup process. If your Raspberry Pi is sitting behind a NAT-based firewall, then you would need to map port UDP 5060 from the router on your public IP address to the private IP address of your Raspberry Pi. In addition, login to your Raspberry Pi as root using SSH and run /root/add-ip to whitelist the public IP address of your 3CX server in the cloud. Otherwise, the 3CX server cannot establish a connection to your Raspberry Pi.
Raspberry Pi Trunk Configuration
Using a browser, login to the web interface for FreePBX on your Raspberry Pi and choose Connectivity → Trunks → Add SIP (chan_sip) Trunk. Name the trunk remote. In the Outgoing Settings, make the entries shown below naming the trunk remote and using a secure secret that will be used to interconnect the two servers. The Register String looks like the following: main:secret@3CX-IP-Address where main is the 3CX server trunk name, secret is your secure secret, and 3CX-IP-Address is the 3CX public IP address.
3CX Trunk Configuration
Using a browser, login to your 3CX server: https://3CX-IP-Address:5001 or http://3CX-IP-Address:5000. From your Dashboard, choose SIP Trunks → Add SIP Trunk. Create a Generic SIP Trunk and then fill in the blanks as shown below. For Registrar/Server/Gateway Hostname or IP, use the public IP address or FQDN of your Raspberry Pi. For Type of Authentication choose Outbound. The authentication credentials should be remote and the secure secret you chose, and the Main Trunk No should match the DID of the Google Voice trunk you set up on your Raspberry Pi. Then pick a default Destination for incoming calls.
3CX Outbound Rules Configuration
Next, we need to tell 3CX which outgoing calls to send out through the Raspberry Pi trunk we just set up. In our example today, we’re going to send all 10-digit calls and 3-digit calls. The 10-digit calls will be routed out the Google Voice trunk on the Raspberry Pi side. And the 3-digit calls will be sent directly to Raspberry Pi extensions. So we’ll need two Outbound Rules.
For the first rule, choose Outbound Rules → Add. For the Rule Name, specify StandardOut. Apply the rule to Calls to Numbers with a length: 10. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.
For the second rule, choose Outbound Rules → Add. For Rule Name, specify StandardInt. Apply the rule to Calls to Numbers with a length: 3. For Route 1, choose Generic SIP Trunk as the Destination. Click OK to save the new rule.
If you already have configured a 3CX smartphone client for one of your 3CX extensions, you now should be able to dial any 3-digit or 10-digit number and have the call processed through your new 3CX→RasPi trunk without any further setup assuming you’ve created a Google Voice trunk on the Raspberry Pi side. That wasn’t too hard, was it?
Routing Incoming Google Voice Calls to 3CX
Depending upon your own requirements, you may want to route incoming Google Voice calls or other trunks directly to an extension and/or softphone on your 3CX server. You obviously could set up multiple trunks of any type on the Raspberry Pi side and have the calls to each trunk routed to a different extension or softphone on the 3CX side. To enable this on the 3CX side, edit your Generic SIP Trunk and click the DIDs tab. Then Add each of the 10-digit DIDs of the Raspberry Pi trunks you wish to redirect. Next, create an Inbound Rule for every DID and tell 3CX where to route the calls.
On the Raspberry Pi side, add each of your Google Voice Trunks. Then create an Inbound Route for each DID and specify the Destination as Trunks → Remote (sip). The 3CX server will take care of routing the various incoming calls to each of the Google Voice trunks to its predefined extension and/or softphone. Enjoy!
Originally published: Monday, March 6, 2017
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
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…
- A simpler Bridge setup is available in the paid versions of 3CX. [↩]
Cellphone Hell: 2017 Minefield Navigation Guide
Well, it’s been an interesting week. RingPlus, a Sprint MVNO, has gone belly up after Sprint pulled the plug on them. Lawsuit details are here. Then, not surprisingly, Sprint announced a new "unlimited" talk, text, and data plan: 5 phones for $90 with a free iPhone with trade-in. After first year, Sprint price escalates to $160 for 4 lines or $190 for 5 lines. And then, Verizon surprised everyone with an "unlimited" plan of their own: 4 phones for $180. With both of these plans, you pay through the nose for the first phone, and then the remaining ones are either free or almost free. So you might as well have some more babies and give them each a phone. For our weary followers that have been with RingPlus, you are about to be introduced to the Sprint Gotcha. Unbeknownst to you, when you inserted that RingPlus SIM and turned on your phone, Sprint locked the phone to their network. And guess what? RingPlus can’t unlock it, and Sprint won’t claiming that you’re not "their customer." But, alas, if you’ve bought your phone, you’re still entitled to use it with a provider of your choice. And, if your phone supports other CDMA carriers such as Verizon or GSM carriers such as AT&T and T-Mobile, you’re in luck. There’s a terrific guy with a company called GSM Zambia, and he will unlock your Sprint phone for $10.84 assuming you have a Windows PC with a USB connector and cable to plug in your phone. For those lucky enough to have a Google-branded phone such as a Nexus or Pixel, you have no worries. Google unlocks it automatically when you insert a SIM card from a different provider.
There are more gotchas awaiting those with iPhones. You see Apple actually makes an iPhone that supports all four of the major U.S. carriers: Verizon, Sprint, AT&T, and T-Mobile. The problem is you probably didn’t get handed that phone. Instead, you got one that was locked to the Sprint network or the AT&T/T-Mobile GSM network, and both of them are missing the necessary radios to support other carriers. But there’s good news. If you’re a loyal customer and have AppleCare for your iPhone, chances are pretty good that Apple will work with you to swap out the phone for one that will work with the carrier of your choice. You have to say this for Apple. Nobody else in the cellphone business would even give you the time of day if you made such a request. So, yes, we are a FanBoy and for very good reason. Apple bends over backwards to help out its loyal customers. Just be advised that you probably will need to speak with an Apple Store manager, and he will probably have to call Cupertino to obtain the document explaining how to handle the transaction. In our case, it was several phones under Apple leases which made things even more complex. But Apple solved it, and they were pleasant about it.
AT&T has had a new "unlimited" plan for about a year, but there were several gotchas in addition to their fine print about what unlimited really means. First, you had to also be a DirecTV customer, but they eliminated that requirement today. And, second, tethering was prohibited. While we’ve previously noted that you could work around the tethering problem by purchasing a ZTE Mobley portable device for your car that could be used outside the vehicle with an adapter. But the wrinkle was AT&T wanted another $40+ a month to cover the device on your unlimited plan. While AT&T boasts that the fourth phone on the unlimited plan is free, it turns out the car device doesn’t meet their definition so, if you only need 3 phones, you still have to cough up the $40 for the mobile device.
T-Mobile also had an "unlimited" plan, but it also restricted tethering. However, T-Mobile is not one to leave money on the table, and they quickly removed the tethering limitation once the Verizon plan was announced. So the bottom line on the 4-phone unlimited plans as of today looks like this: Sprint $90 (10GB tethering), T-Mobile $160 (10GB tethering), AT&T $180 (no tethering), and Verizon $180 (10GB tethering). All four carriers describe their plans as "unlimited" while none truly are insofar as 4G data is concerned. The new buzzword is "deprioritization" which means the carrier reserves the right to slow your data speeds once you reach a certain threshold. Also be advised that zero-rating of certain services is likely to become less of an issue with the Trump administration. In T-Mobile’s case, you get unlimited streaming of certain music and video services at reduced bandwidth. With AT&T, you get streaming of DirecTV movies at reduced bandwidth. With Sprint, you get HD video streaming at no extra cost plus a free iPhone7 for the next 18 months when you trade-in certain older phones. Unless you live in a very busy metropolitan area, user reports suggest that deprioritization shouldn’t be a concern. Here’s the Reddit thread with everything you need to know.
Despite our extreme dislike for almost everything about the Sprint organization and the way they do business, if you happen to live in a city with good Sprint coverage, you really can’t beat their 5 phones for $90 "unlimited" deal at least for one year. After that, Sprint is no bargain at all. If you’re using RingPlus, then that probably means you already have endured Sprint so the change will be easy for you. Just be advised that there are plenty of Sprint reps out there that will try to tell you your phones don’t qualify because they were "prepaid" phones and the plan is only available for "postpaid" phones. A better approach is to visit a Sprint store and advise them that you wish to port your existing phones to the new Sprint unlimited plan. That seems to work although YMMV. Remember, it’s still Sprint you’re dealing with. Good luck!
Feb. 27 Update: The Unlimited Data Plan competition continues to escalate. Today, AT&T sweetened its unlimited plan offering by adding 10GB of free tethering to each phone on its plan beginning Thursday. And T-Mobile announced that customers now can register three phones on its unlimited plan for only $100/month. Unlike Sprint, the T-Mobile offering has no one-year discount cutoff for customers taking advantage of the special pricing. All four major carriers in the U.S. now offer 10GB/month of tethering for each phone on an unlimited data plan.
Published: Thursday, February 16, 2017 Updated: Monday, February 27, 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…
VoIPtopia 2017: Choosing the Best, Free VoIP Platform
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Once a year we like to step back and survey the latest and greatest VoIP developments for the coming year. And 2016 was certainly filled with surprises including the release of free versions of 3CX sporting the PIAF5 and Elastix 5.0 monikers. That, in turn, produced a wave of FUD from our friends at Sangoma® urging users to return to their open source roots. But guess what? Sangoma was pitching their FreePBX Distro®, another closed source product just like 3CX. Sure, the Sangoma distro has open source components… just like 3CX and your car for that matter. But it’s disingenuous to diss other products because they’re closed source platforms when yours is too. So today we want to cut through the sales pitches and compare apples to apples while offering our Elastix friends this New Year’s Day Resolution:
Ignore the Hype! Look Before You Leap and Avoid Jumping from the Kettle into the Fire.
Funny: Sangoma Reinforces Commitment to Open Source by Offering Migration Tool to its Closed Source Distro https://t.co/8QK22cWyi9 #asterisk
— Ward Mundy (@NerdUno) December 29, 2016
NEWS FLASH: For PIAF3 and Incredible PBX users who have registered on the PIAF Forum, you’ll be getting an invitation to upgrade to the 8-simultaneous-call 3CX commercial platform at no cost. In addition to unlimited extensions, this one-year license adds unlimited SIP trunks and gateways, 10-participant conferencing, G.729 support, custom FQDNs, BLF support, Call Parking, Call Queueing, Call Pickup, Call Recordings and Management, Call Reporting, Intercom/Paging, remote 3CX bridging support, as well as an integrated fax server and Office 365 and Microsoft Outlook integration. If you haven’t already joined the PIAF Forum, there’s still time. But you’d better hurry.
Choosing a VoIP platform is partially a subjective decision, but there also are some glaring red flags to consider. We suggest you begin by deciding whether your preferences include any must-have’s. Do your requirements mandate an open source solution? Do you need text-to-speech and voice recognition? Does the platform have to include Asterisk®, or are you open to alternatives? Does the operating system have to be Linux-based and, if so, must it be CentOS, Debian, or Ubuntu? If you’ll be using SIP phones, must the platform include phone provisioning software for your phones, or is the ability to purchase it as an add-on sufficient? Is paid support important in making your platform decision and how much are you prepared to pay? Are automatic or pain-free software updates critical in making your selection? Is migration from an existing platform a factor? Does a preconfigured, secure firewall matter, or are you prepared to do it yourself or take your chances? Before choosing to ignore security, read last month’s RIPS analysis of FreePBX®. Here’s a snippet from the article. Read it carefully. It’s your phone bill.
Since FreePBX is written completely in PHP, we decided to throw it into our code analysis tool RIPS. The results were more than surprising…
The total amount of detected vulnerabilities is very high. Luckily, the majority of the detected vulnerabilities are inside the administration control panel, such that attackers either need to steal a valid account or they have to trick an administrator into visiting a malicious website that triggers one of the critical vulnerabilities. For example, a remote command execution vulnerability could be triggered by a less critical cross-site scripting vulnerability. By chaining both vulnerabilities, the severity is increased drastically and can lead to full server compromise.
In choosing which platforms to include today, we eliminated platforms which we considered too complicated for the average new user to configure. We also eliminated any platform that did not offer at least a free tier of service with a reasonably complete feature set as part of their offering. If we’ve inadvertently missed one of your favorites, please feel free to leave a comment, and we will consider including it as well. Happy Hunting!
VoIP Platform Feature Summary
Aggregation: FreePBX Distro a.k.a. AsteriskNOW
License: Closed Source
VoIP Platform: Asterisk 13/14
GUI: FreePBX GPL and Commercial modules
O/S: CentOS-clone
Phone Provisioning: Open Source (minimal) or Commercial
Text-to-Speech/Voice Recognition: Optional/No
Software Updates: Manual
Migration Tools: Yes
Security: Fail2Ban + User-Configured Firewall
Security Rating (as delivered): see above
Comments: Extensive commercial NagWare preinstalled
Aggregation: Incredible PBX for Wazo
License: GPL3 Open Source
VoIP Platform: Asterisk 14 RealTime
GUI: Wazo GPL3 modules
O/S: Debian 8
Phone Provisioning: Extensive Open Source
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic or 2-minute Manual
Migration Tools: No
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList
Comments: High Availability & Call Center GPL3 Modules
Aggregation: Ombutel
License: Closed Source
VoIP Platform: Asterisk 13
GUI: Ombutel with external module support
O/S: Debian 8
Phone Provisioning: Closed Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Manual
Migration Tools: No
Security: FaiL2Ban + Do-It-Yourself Firewall
Security Rating (as delivered): Insecure
Aggregation: PIAF5 powered by 3CX
License: Closed Source
VoIP Platform: 3CX
GUI: 3CX
O/S: Debian 8
Phone Provisioning: Extensive Closed Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: Yes
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure
Comments: Free upgrade provides unlimited SIP trunks with 8 simultaneous calls
Aggregation: Elastix 5.0 powered by 3CX
License: Closed Source
VoIP Platform: 3CX
GUI: 3CX
O/S: Debian 8
Phone Provisioning: Extensive Closed Source
Software Updates: Semi-Automatic
Migration Tools: Yes
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure
Comments: Free version limited to one SIP trunk & 8 simultaneous calls
Aggregation: Incredible PBX 3
License: GPL2 Open Source
VoIP Platform: Asterisk 13
GUI: FreePBX GPL modules only
O/S: CentOS 6/7, Ubuntu 14, or Raspbian 8
Phone Provisioning: Open Source (minimal)
Text-to-Speech/Voice Recognition: Yes/Yes
Software Updates: Automatic
Migration Tools: Yes
Security: Fail2Ban + Preconfigured Firewall
Security Rating (as delivered): Secure WhiteList
Comments: FreePBX GPL modules only; module signature verification disabled1
Aggregation: Elastix 4.0
License: Open Source GPL
Platform: Asterisk 13
O/S: CentOS 7
Phone Provisioning: Open Source
Text-to-Speech/Voice Recognition: No/No
Software Updates: Semi-Automatic
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Insecure
Comments: Currently unavailable but fork announced
Aggregation: PIAF3
License: Open Source GPL with Closed Source Installer
Platform: Asterisk 11/13
O/S: CentOS 6
Phone Provisioning: Open Source (minimal)
Text-to-Speech/Voice Recognition: No/No
Software Updates: Manual
Migration Tools: No
Security: Fail2Ban + Unconfigured Firewall
Security Rating (as delivered): Insecure
Comments: No longer maintained
Published: Sunday, January 1, 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…
- See RIPStech article explaining why FreePBX module signature verification is a very dangerous methodology. [↩]
If It Walks Like a Duck and Quacks Like a Duck, Guess What?
WOW! When we started our 2016, The Year of (real) VoIP Choice series, little did we know everything that was about to unfold. It’s been an interesting last few months in the VoIP community with the introduction of PIAF5 and Elastix 5.0 and Ombutel and ThirdLane and this week’s XiVO fork to Wazo. But, stay calm. There is a bright light at the end of this tunnel. You now have MORE FREE VoIP PBX CHOICES than ever before. And every one of them is a rock-solid performer. If the word "commercial" sends shivers through your spine, then Ombutel and this week’s new Incredible PBX for Wazo introduction will make this a very bright holiday season for you. If commercial backing with 24/7 support is your cup of tea, ThirdLane’s free offering includes 10 extensions with full product functionality while PIAF5’s free edition includes unlimited extensions with 8 simultaneous calls, a 5-user conferencing module, a SIP trunk of your choice… and No NagWare! 3CX1 also has made a generous offer for those of you that want to start your own business. You can sign up as a reseller, obtain a full NFR product license, and get free training! And, reportedly, a new Asterisk® VoIP Gateway to 3CX is in the works that will let you tie your existing Asterisk-based PBX directly to 3CX giving you the best of both worlds.2 What’s not to like?
We often wonder why more Fortune 500 companies haven’t adopted open source VoIP solutions when their organizations have computer rooms full of Linux servers. If this election season taught us anything, it’s this. You can learn an awful lot about people in just 140 characters. Here’s a snippet of our exchange last week with the Digium® Chief Technology Officer and Sangoma® Vice President which speaks volumes:
This screenshot brought to you from the AsteriskNOW ISO. If it looks proprietary and smells proprietary, guess what? #asterisk pic.twitter.com/YHtaygnvPb
— Ward Mundy (@NerdUno) December 8, 2016
amazing how clueless crazy guy is
— Tony Lewis (@tonyclewis) December 8, 2016
What’s really crazy is these same individuals have no qualms pitching THEIR proprietary software and THEIR proprietary phones while playing dumb. So how do you square the rhetoric with the fact that SwitchVox® AND AsteriskNOW® and the FreePBX Distro® are all closed source ISOs. One has to ask where was the moral outrage when the FreePBX® devs sold out to SchmoozeCom® and then to Sangoma® or when they turned the FreePBX ISO into a closed source product. That, of course, was different because it was money in their pockets, not to mention cushy new full-time jobs singing the praises of "open source." But nobody wants to talk about any of that. In the real estate business, these guys are called NIMBYs, an acronym for "Not In My Back Yard." They’re all for change as long as it doesn’t affect their own neighborhood and pocketbook. To translate it into VoIP-speak, these are the folks that would prefer you stick with THEIR code generator and buy boatloads of THEIR commercial, closed source modules and THEIR proprietary phones. To everyone else, keep off our playground! Make no mistake. It’s all about the money!
Not surprisingly, a virtually identical feature set is provided at no cost on the ThirdLane and 3CX platforms. So be sure to compare apples to apples and ignore the rants. After all, IT’S YOUR CHOICE. Kick the tires of all the products and choose the platform that best meets your needs and those of your organization. I’m reminded of an old legal adage: "When the facts are on your side, pound the facts. When the law is on your side, pound the law. And when neither is on your side, pound the table." Those that want to distract you from considering the merits of other products by launching attacks on their competitors are little more than table pounders. So consider the source especially when some of the loudest and most vocal members of the fan club are on the payroll hiding behind a cloak of anonymity. None are innocent bystanders. It’s all about the money!
So… are there any Asterisk®-based products that really are released under an open source license? Actually, there are several. The Incredible PBX platforms for CentOS, Ubuntu, and Raspbian as well as the Incredible PBX 13 ISO are all open source products that include the latest LTS version of Asterisk. And then there’s Incredible PBX for XiVO and (NOW!) Wazo, two virtually identical GPL3 platforms that feature an Asterisk real time environment with a more sophisticated GUI and full API support. We’ll have more to say about the latest Wazo release featuring Asterisk 14 later this week. Stay tuned!
Why Incredible PBX? Glad you asked. Here’s my short answer from the PIAF Forum:
The inspiration for Incredible PBX was to save people the unbelievably steep learning curve we endured when first starting to use Asterisk over a decade ago. And, frankly, the developers liked it that way because many of them made a living configuring Asterisk for people that didn’t know what they were doing.
What you get with Incredible PBX?
- You get a secure server out of the starting gate unlike any other distro.
- You get all the tools and samples to learn how to do anything with Asterisk.
- You get a working system out of the box that can make and receive FREE calls.
- You get a pure open source GPL platform with No Gotchas and No NagWare.
What you don’t get with Incredible PBX?
A college degree in telecommunications or network administration without actually doing the work. Yes, it’s hard. But, with Incredible PBX, it can also be fun AND safe.
Published: Monday, December 12, 2016
Need help with Asterisk? Visit the PBX in a Flash Forum.
Coming Soon to Nerd Vittles: The Autonomous Car
Longer version of self-driving demo with Paint It Black soundtrack https://t.co/YuUmyEaCgR
— Elon Musk (@elonmusk) November 19, 2016
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…
[VoIPtopia: Google Services with Incredible PBX and PIAF5
It’s been a while since we provided a fresh look at Google Voice, Google SMS messaging, and Google’s Speech Recognition labyrinth which have been integral components of Incredible PBX for many years. For those living in the United States, here’s a soup-to-nuts tutorial to get all of the services deployed quickly on any Incredible PBX platform including XiVO and Elastix as well as on the new freeware releases of Ombutel and PBX in a Flash 5 powered by 3CX. On most of the platforms, you can deploy Google Voice services directly; however, with PIAF5, Elastix, and Ombutel you’ll need to set up a SIP trunk using the Simonics SIP to Google Voice gateway to take advantage of free calling in the U.S. and Canada with Google Voice.
Implementing Google Voice with Incredible PBX
Before you can obtain Google Voice service to make free calls in the U.S. and Canada, you’ll need several things: (1) a Google account, (2) access to a computer with an IP address in the United States, and (3) a U.S. phone number to verify your residence for Google Voice.
To get started, sign up for a Gmail account here:
Once your Gmail account is created, click Allow and then Allow and Remember when prompted whether to Allow Gmail to run "Google Talk."
In a separate tab of the same browser, go to Google Voice to sign up for an account. Begin by choosing whether to obtain a new phone number for Google Voice or whether you wish to use an existing mobile phone number that you already own. Next, choose a forwarding phone number which will ring when your Google Voice number is called. NOTE: You do not need to keep this activated on your account once it is completely set up. Be advised that Google also plays games with certain phone numbers such as pretending to ring them when, in fact, they haven’t placed a verification call at all. This usually is because of prior abuse of the number with the Google Voice service or because you’ve gotten greedy and signed up for too many free numbers. If a number doesn’t work for verification, you’ll need to choose another number. And it’s usually a good idea to create additional Google Voice accounts from different IP addresses. Once you complete the verification step, you can choose a phone number in an area code of your choice. Same thing holds for picking phone numbers. If you get error messages saying to "try later," what Google is really telling you is you’re a greedy bastard. Set up additional Google Voice numbers from a different computer using a different IP address and chances are the problem will go away. It did for us. 😉
Once you have your new Google Voice number, Google will drop you into the Voice Inbox. Ignore offers to activate, enable, or do anything else with Hangouts. Otherwise, you may kill the ability to use your new Google Voice number with Asterisk®.
Click on the Settings Gear icon in the upper-right corner of the window. In the Phones tab, make certain that (1) Google Chat is enabled and (2) your forwarding phone number is disabled:
In the Calls tab, make it look like this for proper Google Voice operation with Asterisk:
You now have a basic Google Voice setup on the Google side to support Asterisk calling. But the default setup uses plain-text passwords for your Google Voice account, and this is not only a security issue, but it also will cause problems if you move your Google Voice account to a different computer. For that reason, we strongly recommend setting up OAuth 2 authentication for your Google Voice account.
Obtaining an OAuth 2 Token for Google Voice
To deploy Google Voice with OAuth 2 authentication on the Incredible PBX platforms that support direct connections to Google Voice (Incredible PBX 13 and Incredible PBX for XiVO), you will first need to obtain an OAuth 2 Refresh Token from Google. On the remaining platforms that require a SIP account using the Simonics SIP to Google Voice gateway (PIAF5 powered by 3CX, Ombutel, and Elastix), you can skip this section since the Simonics site will obtain the refresh token for you as part of the signup process.
While you’re still logged into your Google Voice account, you need to obtain a refresh_token which is what you’ll use instead of a password when setting up your Google Voice accounts with Incredible PBX 13 and Incredible PBX for XiVO. Here’s how.
1. Be sure you are still logged into your Google Voice account. If not, log back in at https://www.google.com/voice.
2. Go to the Google OAUTH Playground using your browser while still logged into your Google Voice account.
3. Once logged in to Google OAUTH Playground, click on the Gear icon in upper right corner (as shown below).
3a. Check the box: Use your own OAuth credentials
3b. Enter Incredible PBX OAuth Client ID:
466295438629-prpknsovs0b8gjfcrs0sn04s9hgn8j3d.apps.googleusercontent.com
3c. Enter Incredible PBX OAuth Client secret: 4ewzJaCx275clcT4i4Hfxqo2
3d. Click Close
4. Click Step 1: Select and Authorize APIs (as shown below)
4a. In OAUTH Scope field, enter: https://www.googleapis.com/auth/googletalk
4b. Click Authorize APIs (blue) button.
5. Click Step 2: Exchange authorization code for tokens
5a. Click Exchange authorization code for tokens (blue) button
5b. When the tokens have been generated, Step 2 will close.
6. Reopen Step 2 and copy your Refresh_Token. This is the "password" you will need to enter (together with your Gmail account name and 10-digit GV phone number) when you add your GV trunk in Incredible PBX 13 GUI. On the XiVO platform, log into your server as root and run: /root/add-gvtrunk. Store this refresh_token in a safe place. Google doesn’t permanently store it!
7. Authorization tokens NEVER expire! If you ever need to remove your authorization tokens, go here and delete Incredible PBX Google Voice OAUTH entry by clicking on it and choosing DELETE option.
Switch back to your Gmail account and click on the Phone icon at the bottom of the window to place one test call. Once you successfully place a call, you can log out of Google Voice and Gmail.
Yes, this is a convoluted process. Setting up a secure computing environment often is. Just follow the steps and don’t skip any. It’s easy once you get the hang of it. Sleep well.
Configuring Google Voice Trunks with Incredible PBX
The setup procedure differs a bit with Incredible PBX for XiVO and Incredible PBX 13.
With Incredible PBX for XiVO, log into the Linux CLI with your root credentials and run: /root/add-gvtrunk. Enter your Google email address, refresh token, and 10-digit Google Voice number when prompted. Follow the instructions which appear when the script finishes, and you’ll have a functioning Google Voice trunk in less than a minute.
With Incredible PBX 13, log into the Incredible GUI as admin using a web browser. Choose Connectivity -> Google Voice -> Add Account and fill in your Google Username, Refresh Token, and 10-digit Phone Number. Check the Add Trunk and Add Outbound Route check boxes. Then click Submit. Create an Inbound Route to tell Asterisk how to route incoming calls to your 10-digit DID. Finally, log into the Linux CLI as root and restart Asterisk: amportal restart.
Simonics SIP to Google Voice Gateway Setup
There’s a one-time fee of $4.99 to use the Simonics gateway if you take advantage of the Nerd Vittles signup link. All remaining Google services are free. You obviously can use the Simonics gateway with almost any PBX that supports SIP trunks, but it’s particularly well-suited for PBXs that don’t natively support Google Voice with OAuth 2 authentication such as PIAF5, Ombutel, and Elastix. To get started, you’ll need to set up an account at Simonics using your existing Google Voice credentials.
1. Using your favorite browser, log in to the Google Voice account you wish to associate with the Simonics SIP gateway. Be sure that you’ve enabled Google Chat in your Google Voice setup.
2. Using a separate tab of your browser, connect to the Simonics Google Voice Gateway site.
3. Go through the steps to register your Google Voice account with the Simonics Google Voice gateway and obtain your credentials.
4. For those using PIAF5, Ombutel, or Elastix, use another tab of your browser to open the GUI interface and create a new SIP trunk using your new Simonics SIP login credentials. With Incredible PBX 13, here’s the drill. Replace 8005551212 with your actual Google Voice number and YOUR-SIP-PW with your actual Simonics SIP password in BOTH the PEER Details and Registration String. Add your Google Voice number to the end of the Registration String like this: GV18005551212:YOUR-SIP-PW@gvgw.simonics.com/8005551212
5. Regardless of PBX platform, the next step is to create an Inbound Route for your incoming calls using either your Simonics username or the 10-digit number you entered at the end of the Registration String in step #4a. This obviously depends upon your PBX platform.
6. Create an Outbound Route for outgoing calls that should be handled by your Google Voice trunk. The CallerID number will be your Google Voice number. You cannot change it.
7. If you’d prefer to send incoming calls from the Simonics gateway to a designated SIP URI instead of the server that registered with the Simonics gateway, enter the address in the format: pbx@myserver.xyz. For additional details, read our previous article on SIP URIs.
SMS Messaging with Google Voice
On the Incredible PBX 13 and Incredible PBX for XiVO platforms, the python setup to support SMS messaging through Google Voice is already installed. On the PIAF5, Ombutel, and Elastix platforms, you’ll first need to install it. Here’s how.
Log into your server as root using SSH or Putty and issue the following commands to install the Google Voice CLI tools:
cd /root apt-get -y install python-setuptools wget http://incrediblepbx.com/install-gv-cli chmod +x install-gv-cli ./install-gv-cli
Before the SMS messaging tools will work, there are two preliminary steps that you must complete on every platform. This is because SMS messaging with python uses plain-text passwords for Google Voice, and Google imposes new hoops that you must jump through in order to continue to use such passwords. While logged into your Google Voice account with a browser, click on this link to Enable Less Secure Apps. Next, click on this link to Activate the Google Reset Procedure. You now have a couple of minutes to actually connect to your Google Voice account from your new server using plain text passwords. This will WhiteList the IP address of your server. So let’s send an SMS message quickly so that everything gets squared away.
To Send an SMS Message Blast to one or more destinations, (1) create a message in /root/smsmsg.txt, (2) specify the SMS numbers in /root/smslist.txt, (3) insert your Google credentials (using your plaintext Google Voice password) in /root/smsblast, and (4) run /root/smsblast to send the message.
Implementing Google’s Speech Recognition API
Speech Recognition currently works with Incredible PBX 13 and Incredible PBX for XiVO only. But we’ll be collaborating with the 3CX folks to bring it to their platform soon. All of the necessary components to use speech recognition for voice dialing from the AsteriDex phonebook (411) and to take advantage of the Siri-like Wolfram Alpha service (4747) already are in place with Incredible PBX 13. While voice dialing works great with XiVO, Wolfram Alpha is just around the corner on the XiVO platform. Before you can actually use voice recognition, you’ll need a Google API key since Google handles the speech-to-text translation on the Asterisk platform thanks to Lefteris Zafiris’ terrific speech-recog AGI script. Here’s a revised step-by-step tutorial to get your API key from Google and activate it on your PBX.
Place a test call by dialing 4-1-1 and saying "Delta Airlines" when prompted. You should be connected to Delta’s reservation system. Enjoy!
Published: Monday, November 14, 2016
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 Stealth AutoAttendant for Incredible PBX and PIAF5
This week we’re dusting off an oldie but goodie, The Stealth AutoAttendant. If you missed our original column 8 years ago, here’s a quick refresher. When a call comes into your PBX, a generic greeting is played: "Thanks for calling. Please hold a moment while we locate someone to take your call." Then the call is transferred to an extension or ring group. Stealth comes into play because this is really an AutoAttendant and, while the greeting is played, a caller can press a preassigned key to transfer the call to some other destination. While it’s obviously not a secure method for providing additional phone features to certain callers, it’s nevertheless helpful in opening up additional PBX functionality without making callers feel like they’re dealing with yet another IVR when they call your home or office. Using the 12-button keypad and clever design, features such as conferencing and DISA can be offered while still providing security through added prompts for passwords or PINs.
https://youtu.be/dB-Tr3WnrKE
In addition to releasing GPL voice prompts from Allison Smith, today we’ll show you how to implement this on three different platforms: Incredible PBX 13 with its FreePBX® GPL modules, Incredible PBX for XiVO®, and PBX in a Flash 5 powered by 3CX®.
Stealth AutoAttendant Voice Prompts
Let’s start by getting you the voice prompts to support the Stealth AutoAttendant. What’s included is a copy of the GPL3 license and versions of the voice prompts in both GSM and WAV format. The prompts are identical except one has a two-second pause at the beginning of the prompt. This is helpful to avoid premature playing of the voice prompt when using trunk providers such as Google Voice that take a couple seconds to set up audio on a call. You can experiment with both and see which best meets your own requirements.
For the techies, it’s probably worth documenting how these prompts were created. We started with GSM versions of both our Generic Greeting prompt and the 2-second silence prompt that can be found on most Asterisk® servers. These were then converted to WAV format: sox 2.gsm -s -b 16 2.wav. Then the WAV files were combined like this:
sox -m 2.wav nv-GenericWelcome.wav nv-GenericWelcome2.wav
Now let’s download the Stealth AutoAttendant prompts to your desktop. The files are available in stealth.zip and stealth.tar.gz format so choose the download that is easiest for you to work with. Once downloaded, unzip or untar the compressed files.
Stealth AutoAttendant with Incredible PBX 13
On the Incredible PBX 13 platform with its FreePBX GPL modules, there are 3 steps to implement the Stealth AutoAttendant. The tricky part in setting up the Stealth AutoAttendant is getting the prompt installed so that it can be used as an Announcement when building an IVR in the GUI. So let’s start there.
1. FreePBX provides a facility for importing existing voice prompts in the GUI. Login as admin to get started. Then go to Admin → System Recordings → Upload and import the generic greeting desired from your desktop. Name the recording and then click Save to add it to your server.
2. Next choose Application → IVR and create a new IVR using the generic greeting as your Announcement. You can follow the IVR Demo template provided with Incredible PBX if you need some hints on how to set up an IVR. Or, better yet, review the Nerd Vittles IVR tutorial.
3. The final step is to point the Inbound Route for one or more of your DIDs to the Stealth AutoAttendant to take incoming calls. Go to Connectivity → Inbound Routes and set the Destination for the DID to the IVR you just created.
Stealth AutoAttendant with Incredible PBX for XiVO
For those using the Incredible PBX for XiVO platform, the voice prompt for the Stealth AutoAttendant already is in place in /usr/local/share/asterisk/sounds. Just copy it to /var/lib/xivo/sounds/playback in order to use it as a voice prompt with your IVRs:
cp -p /usr/local/share/asterisk/sounds/nv-GenericWelcome.wav \\ /var/lib/xivo/sounds/playback/.
Currently, there is no GUI to create IVRs and AutoAttendants with XiVO, but an IVR GUI is in the works so stay tuned. In the meantime, our tutorial and IVR template (ivr-1.conf in /etc/asterisk/extensions_extra.d) will show you how easy it is to create the dialplan code. Just copy it (cp -p) to stealth-aa.conf as a starting point. To use the Stealth AutoAttendant voice prompt, simply change the third line from ivr-Allison and replace it with the new voice prompt: nv-GenericWelcome. It doesn’t get much easier than that.
Finally, using the XiVO GUI, navigate to IPBX → Call Management → Incoming Calls and create a new incoming route for your DID that points to the IVR template you created.
Stealth AutoAttendant with PIAF5 powered by 3CX
On the PIAF5 and 3CX platform, implementing the Stealth AutoAttendant is the easiest of all. It’s all done on a single screen. You get what you pay for. 🙂 Start by logging into the 3CX web portal. When the Dashboard appears, click on Digital Receptionist in the left toolbar. Next click Add. 3CX will automatically assign an extension number for your new IVR, e.g. 800. Assign a new friendly name: Stealth AutoAttendant. Then click the Upload button to upload the WAV prompt from your desktop. Choose the options desired for your Stealth AutoAttendant and designate the DID(s) to use for incoming calls to the Stealth AutoAttendant. In the Destination for no or invalid input, set the Timeout to 2 seconds and choose where to route incoming calls when the caller presses no keys or when the input is invalid. This means the caller has the duration of the greeting plus 2 seconds to press a key and divert the call to another destination. Click OK to save your settings. Told you it’s easy.
Published: Monday, October 31, 2016
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…
Take the XiVO Plunge: 4 Months of Free Cloud Hosting
Nobody has to tell us how painful change can be. We oversaw the deployment of over 30,000 IBM PCs only to switch horses and become a dedicated Mac lover. And we’ve invested almost 10 years in another Asterisk® GUI only to be disappointed by the direction of that project. That led to our New Year’s Resolution to find a better mousetrap for unified communications open source development. And, boy, did we find one. So here’s the deal. You either believe in the open source community and want to foster free and open development of software, or you don’t. And, if you don’t, that’s perfectly fine. There are lots of commercial PBX alternatives including the terrific 3CX products from our platinum sponsor. But don’t wrap yourself in the open source flag, brag about free and freedom, and then market a product that is none of the above. If your distro’s license agreement prohibits redistribution thereby discouraging sharing which is the lynchpin of the GPL, then the product has little if anything to do with free and freedom.
The good news is we’ve now found an awesome alternative that is pure open source code with an actual GPL3 license. So come join the party and lend a hand with your suggestions and/or your code contributions. We’ll put your name in bright lights, and the open source community will be forever in your debt. Our challenge is to get you as excited about XiVO as we are. There’s nothing with VoIP and Unified Communications that you can’t do better, cheaper, and faster using XiVO. And XiVO’s Asterisk RealTime implementation has no competition, period. Instead of lengthy delays to process changes, rewrite Asterisk config files, and reload the entire Asterisk dial plan, Asterisk RealTime brings instantaneous configuration updates.
We can think of no better way to introduce you to this terrific platform than offering up a free cloud platform until 2017 to let you kick the tires. It won’t impact your production servers while letting you explore the possibilities offered by a state-of-the-art Asterisk 13 platform with no equal. Believe me. We know every wart and pimple in the old GUI platform, and you won’t have to wrestle with any of the traditional problems that we all assumed were native to Asterisk. Guess what? They weren’t. No, your server won’t blow up when you add a new module. No, Asterisk won’t refuse to start because you chose to upgrade an existing component. No, you won’t be Nickle and Dimed into buying critical platform enhancements. And, no, you won’t be charged hundreds of dollars for "support" only to be told that you need to switch to a more proprietary platform. Yes, the XiVO development team releases seamless upgrades every three weeks at no cost. Yes, uncrippled endpoint provisioning for dozens of phones is provided in XiVO at no cost. Yes, powerful call center and High Availability technology is included at no cost. And, yes, backups of your server are made every night for free.
There’s more good news. VULTR is a relatively new cloud provider that now hosts virtual machines in over a dozen cities around the world. For new subscribers, they are offering a $20 credit when you sign up using our referral link. And, yes, your registration provides a few shekels to Nerd Vittles to keep the lights on. The great news is that $20 buys you a full four months of XiVO cloud hosting service, and you won’t find a better do-it-yourself platform at any price, let alone free.
Building the Debian 8 Platform at Vultr for XiVO
The first step in your XiVO adventure is to sign up for a Vultr account with your $20 credit using the Nerd Vittles referral link. Once you’ve done that, it’s time to build your Debian 8 virtual machine to host XiVO in the Cloud. (1) Choose your favorite city to host your server, (2) pick the Debian 8 64-bit platform, and (3) choose the $5/month server size.
IMPORTANT: Leave the Server Hostname & Label blank!
Once your virtual machine is up and running, log in with SSH or Putty using the root password provided. Do NOT install XiVO from the console, or the firewall will lock you out of your own machine! Change your root password immediately: passwd.
Next, set up a swap file on your virtual machine, or the XiVO install will fail on the $5 platform:
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
Installing Incredible PBX for XiVO in the Vultr Cloud
While still logged into your server as root using SSH/Putty, issue the following commands to kick off the install:
cd /root wget http://incrediblepbx.com/IncrediblePBX13-XiVO.sh chmod +x IncrediblePBX13-XiVO.sh ./IncrediblePBX13-XiVO.sh
The initial setup brings your Debian 8 server up to current specs, and then the virtual machine will reboot. After rebooting, log into your server again as root with your new root password. Issue the following command to complete the XiVO and Incredible PBX installation and configuration:
./IncrediblePBX13-XiVO.sh
You’ll be prompted to set your time zone, passwords, and choose the optional features of Incredible PBX you wish to install. We strongly recommend you install ALL of the Incredible PBX feature set. Many cannot be added later.
Verify that the XiVO install completed successfully when prompted. Then verify that the XiVO initial configuration completed successfully by once again pressing ENTER. The firewall and Incredible PBX install will then proceed without further prompting. Total setup time: under 10 minutes.
There still are some setup steps required, and these are performed within the XiVO GUI using a web browser. For step-by-step instructions on the Incredible PBX Initial Configuration Procedure, click here. Enjoy your adventure!
Originally published: Monday, July 25, 2016
9 Countries Have Never Visited Nerd Vittles. Got a Friend in Any of Them https://t.co/wMfmlhiQ9y #asterisk #freepbx pic.twitter.com/TPFGZbqWB6
— Ward Mundy (@NerdUno) April 22, 2016
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…
Security 101: A Fresh Look at Incredible PBX Security Audit Methodology
Incredible PBX remains one of the most secure VoIP server platforms on the planet for one simple reason. We always deploy a preconfigured Linux IPtables firewall with a whitelist that hides your server from everyone except you and trusted VoIP providers. IPtables is automatically configured and deployed as part of every initial install of Incredible PBX regardless of your platform. This includes XiVO with Debian 8 as well as CentOS 6 and 7, Ubuntu 14.04, Raspbian 7 and 8, and even SHMZ OS (not recommended). If your server happens to be housed behind a hardware-based firewall as well, then so much the better. That obviously isn’t possible with most Cloud-based servers so IPtables firewall security is a must.
Unlike most other VoIP server platforms, we don’t leave firewall configuration to chance. Nor do we assume you’re a firewall expert. It really doesn’t matter whether you are or not, you still need a server platform that is secure and protected. So we do it for you initially and, if you are a firewall expert or study to become one, you then can modify the default settings to meet your own requirements down the road. In the meantime, you and your server are protected.
As you probably have surmised, we conduct periodic security audits of our servers testing for vulnerabilities. And we perform these audits locally as well as remotely using servers we’ve deployed throughout the world. We also deploy honeypot servers from time to time in order to gather important information about what the bad guys are up to. With as many platforms as Incredible PBX now supports, just conducting local and remote security audits is no small feat.
Today we want to share some of the methodology we use in conducting our audits, and we’ll provide the results of our most recent remote security audit. We encourage everyone with a VoIP server, whether it’s Incredible PBX or some other platform, to periodically test your server(s) for vulnerabilities AND access. It not only could save you thousands of dollars, but it also protects the rest of us by assuring that you haven’t inadvertently provided malicious individuals with a zombie platform from which to launch denial of service and spam attacks against the Internet community. So let’s get started.
The first step in testing your server is to log into your server as root using SSH or Putty from multiple IP addresses. These sites should include logins from the home base of your server if it’s a dedicated machine, from your home PC, from a neighbor’s PC, from a public WiFi hotspot, and from your smartphone as well as someone else’s. If you gain access from all of these sites, you’ve got a problem. It means SSH access is not protected in any way on your server. While SSH is relatively secure, it has had its share of problems. And zero day vulnerabilities are regularly discovered in various Linux utilities so exposing all of your server’s important resources to the Internet is a very bad idea.
The second test deciphers the existing firewall rules that have been activated on your server: iptables -nL. If the results look like the following, you’ve got a major problem. It means there are no firewall rules blocking any access to your server:
root@incrediblepbx:~ $ iptables -nL Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination
Next, reboot your server and repeat the first two tests to make certain that your firewall still is activated properly whenever your server experiences a power outage and comes back on line.
If your firewall is not running, try issuing the command, iptables-restart, and then retest: iptables -nL. If you get the same results shown above, then something has come unglued. Here’s how to easily fix things up. First, move to the directory where the iptables rules are stored on your server. For CentOS/SL/RHEL, it’s /etc/sysconfig. For Debian/Ubuntu/Raspbian, it’s /etc/iptables.
Next, copy the default Incredible PBX firewall settings to the proper file location.
For CentOS/SL/RHEL platforms:
cp -p /etc/sysconfig/rules.v4.ubuntu14 /etc/sysconfig/iptables cp -p /etc/sysconfig/rules.v6.ubuntu14 /etc/sysconfig/ip6tables
For Debian/Ubuntu/Raspbian platforms:
cp -p /etc/iptables/rules.v4.ubuntu14 /etc/iptables/rules.v4 cp -p /etc/iptables/rules.v6.ubuntu14 /etc/iptables/rules.v6
Next, edit iptables (CentOS/SL/RHEL) or rules.v4 (Debian/Ubuntu/Raspbian) and move to the bottom of the file where you’ll find a section that looks like this:
# The IP addresses are your server, user, and public addresses respectively -A INPUT -s 8.8.4.4 -j ACCEPT -A INPUT -s 8.8.8.8 -j ACCEPT -A INPUT -s 74.86.213.25 -j ACCEPT
Replace the existing IP addresses with the actual IP addresses of your server, user workstation, and public IP address. Be very careful here. If you don’t whitelist the IP address of the machine on which you are performing these tasks, you will lock yourself out when you restart your firewall. Once you’ve made the changes, save the file.
Finally, restart IPtables using the following command: iptables-restart. Then retest: iptables -nL.
We’re not going to spend a lot of time addressing what the proper firewall rules for your VoIP server should be. If you’re interested, you can take a look at the IPtables firewall setup that is deployed with Incredible PBX. On RHEL/CentOS/SL servers, you’ll find the firewall rules in /etc/sysconfig/iptables. On Debian/Ubuntu/Raspbian servers, the rules are in /etc/iptables/rules.v4. Suffice it to say that, if the only remote access required with your server is to connect to VoIP service providers, there is no reason to expose your web server or your SIP ports to the Internet, period. And this is true whether your server is sitting behind a hardware-based firewall or not.
The Incredible PBX security design uses a whitelist to provide access to most network services other than those that are absolutely essential to the operation of your server. The reason we use a whitelist is because blacklists don’t work. Those interested in doing harm to your server are perfectly capable of altering their IP addresses until they find one that isn’t blacklisted. And they also are adept at poisoning blacklists with IP addresses that are absolutely essential to the operation of your server, e.g. DNS servers and NTP servers.
As part of every Incredible PBX firewall install, we provide SIP and IAX access to many of the major VoIP providers around the globe. You may be wondering why we use IP addresses for providers rather than fully-qualified domain names. The reason is that IPtables doesn’t directly support FQDNs. Instead, when IPtables starts up, it looks up every FQDN and converts it into an IP address. If a server matching the FQDN happens to be off line, IPtables crashes and burns. The same is true if the lookup is attempted before DNS services are running on your server. So, the short answer to why we use IP addresses is because it is safer. The downside, of course, is you can’t eyeball the IP address and decipher to whom it belongs. If you ever have any doubt about the identity of the provider associated with any specific IP address, there’s a simple utility you can run to identify its owner: nslookup 178.63.143.236.
Here is a list of the providers included in the default Incredible PBX whitelist. Others can be added using the add-ip and add-fqdn utilities in /root. If you use FQDNs, be sure to add the entries to /root/ipchecker so that your IP addresses are periodically checked and updated when necessary. This is especially important for dynamic IP addresses at remote locations.
outbound1.vitelity.net inbound1.vitelity.net atlanta.voip.ms chicago.voip.ms dallas.voip.ms houston.voip.ms losangeles.voip.ms newyork.voip.ms seattle.voip.ms tampa.voip.ms montreal.voip.ms montreal2.voip.ms toronto.voip.ms toronto2.voip.ms london.voip.ms didforsale.com callcentric.com sipgate.com chi-in.voipstreet.com did.voip.les.net magnum.axvoice.com proxy.sipthor.net sip.voipwelcome.com incoming.future-nine.com outgoing.future-nine.com DEN.teliax.net LAX.teliax.net NYC.teliax.net ATL.teliax.net IPkall (defunct) used two IP addresses: 66.54.140.46 and 66.54.140.47 gvgw1.simonics.com sip2sip.info googlelabs.com talk.google.com gmail.com
The major drawbacks to firewall whitelists are (1) you can inadvertently lock yourself out of your own server and (2) someone that needs access to your server from remote locations may have more difficulty connecting without intervention by a network administrator to authorize remote access. With Incredible PBX, we’ve provided some tools to ease the pain. First, Incredible PBX is deployed with both the PPTP and NeoRouter VPN platforms already in place. With a VPN IP address, remote logins are minimized because they work from almost anywhere. Second, Incredible PBX includes the PortKnocker utility which lets a remote user "knock" on the server using three randomly assigned port numbers to gain temporary access. Many Incredible PBX platforms also support Travelin’ Man 4 which lets you authorize remote access by telephone. You also need to test remote VPN, PortKnocker, and Travelin’ Man 4 access as part of your security audits.
Testing for vulnerabilities is only half of the puzzle. Also make certain that your server has the proper Linux tools in place to allow you to whitelist additional IP addresses so that remote users can deploy phones or gain access to your server when necessary. Try to run the nslookup and dig utilities to verify that they are installed on your server. If not, install them with yum install bind-utils (CentOS/SL/RHEL) or apt-get install dnsutils (Debian/Ubuntu/Raspbian).
Security Audit Results. We’re pleased to report that no vulnerabilities were identified in any of the Incredible PBX platforms; however, good security practices dictate that the IPkall IP addresses should probably be removed from the whitelist now that the company has ceased providing VoIP services.
For CentOS/SL/RHEL platforms:
sed -i '/66.54.140.46/d' /etc/sysconfig/iptables sed -i '/66.54.140.47/d' /etc/sysconfig/iptables sed -i '/66.54.140.46/d' /etc/sysconfig/rules.v4.ubuntu14 sed -i '/66.54.140.47/d' /etc/sysconfig/rules.v4.ubuntu14 iptables-restart
For Debian/Ubuntu/Raspbian platforms:
sed -i '/66.54.140.46/d' /etc/iptables/rules.v4 sed -i '/66.54.140.47/d' /etc/iptables/rules.v4 sed -i '/66.54.140.46/d' /etc/iptables/rules.v4.ubuntu14 sed -i '/66.54.140.47/d' /etc/iptables/rules.v4.ubuntu14 iptables-restart
We did identify a couple of access anomalies that kept the add-ip and add-fqdn utilities in /root from functioning properly. These glitches meant that a few administrators could not easily add remote IP addresses to their whitelists. Three fixes are recommended. First, be sure the utilities documented in the previous paragraph are installed on your server. Second, on CentOS/SL/RHEL platforms or servers installed using the Incredible PBX ISO, issue the following commands after logging into your server as root:
sed -i 's|/etc/iptables/rules.v4|/etc/sysconfig/iptables|' /root/add-ip sed -i 's|/etc/iptables/rules.v4|/etc/sysconfig/iptables|' /root/add-fqdn
Third, for Incredible PBX deployments on the CentOS 7 platform, issue these commands while logged in as root:
chattr -i /root/add-ip sed -i 's|iptables-persistent|iptables|' /root/add-ip chattr +i /root/add-ip
Be safe!
Originally published: Tuesday, August 9, 2016
Go For the Gold: Incredible PBX for XiVO, CentOS, Ubuntu, or Raspbian. https://t.co/zZDnO8OUVt #asterisk #NoGotchas pic.twitter.com/wv3KkuMJha
— Ward Mundy (@NerdUno) August 8, 2016
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…
Free Asterisk Solutions
