Posts tagged: voip

Incredible PBX on Steroids: The Asterisk-GUI Pilgrimage Begins (Chapter 1)

As the holiday season gets underway with Thanksgiving, Hanukkah, Christmas, and especially Festivus, we thought it might be interesting to actually provide a running dialog of how a new Asterisk® project is born and what hurdles and solutions are encountered along the way. We mentioned last week that we were dusting off Mark Spencer’s Asterisk-GUI with hopes of transforming it into an updated Asterisk 11 platform for hobbyists and SOHO telephony users with many of the ease-of-use touches that have made Incredible PBX a big hit. So today we officially kick off the adventure with a look back at Week One. Our target, by the way, is a New Year’s Day release to celebrate the arrival of 2015.

This is the first installment in our series. You can catch up with the Overview as well as Chapter 2 and Chapter 3 here.

Project Development Roadmap

You may be asking, “What’s in it for me?” Well, lots! One of the unfortunate side effects of having always relied upon the FreePBX® GUI for Asterisk administration is you never really learned how Asterisk works. Nor did we ever quite appreciate its lightning-fast performance. We’re as guilty as anyone for over-reliance on a design tool without much appreciation for its interaction with the actual communications server. And, like many things in life, you form some bad habits along the way that are hard to break. Don’t get us wrong. There are thousands of things to like about FreePBX and, for production-level servers hosting dozens or hundreds of users, it remains a very comfortable choice and our hands-down favorite.

We resolved early on to approach the Asterisk-GUI remake a little differently. We plan to actually document why we’re going down certain paths and what the benefits will be for the ultimate user. There won’t be any convoluted code to deter your learning how things actually work. And there won’t be any patent, trademark, or copyright gotchas to hinder your forking or repurposing our code to meet your own requirements. And, finally, there won’t be any license fees, hidden or otherwise. Just comply with the GPL2 license as written and be our guest! From our vantage point, that’s what open source is all about.

Defining Project Objectives

We began the week by sketching out some objectives as well as defining some likes and dislikes. As we mentioned last week, the objective is not to replace FreePBX for those that actually need that horsepower. First and foremost we want to design this product for the target audience: hobbyists, home users, and SOHO businesses. Many of the platforms we are targeting have limited memory and only modest computational ability. Many of the people in the target audience have never used a PBX before and know little to nothing about networks and security. We don’t want anyone blindsided by a $100,000 phone bill because they didn’t know how to implement a firewall so we’ll include a preconfigured one as part of the install. And, like all Incredible PBX systems, an automatic update utility will be included to keep your system current AND safe!

Second, we wanted a product that was incredibly simple to put into production. Ease of configuration was a definite must-have. With many GUIs (think: Microsoft Windows), developers get so enamored with the brilliance of their own creation that they lose sight of the fact that typing a short list of usernames and passwords often is much simpler than navigating through dozens of data entry screens with hundreds of mouse clicks to enter the same information.

We also are steering clear of reinventing the proverbial wheel. Mark Spencer and his colleagues are some of the most talented programmers on the planet. To the extent that the original, feature-rich Asterisk-GUI creation can be implemented without major plumbing changes, that is not only desirable but absolutely essential in bringing this new product to market within weeks, not months or years.

Keep in mind that both FreePBX and Asterisk-GUI are code generators for Asterisk. No call is actually processed by FreePBX or Asterisk-GUI. From a system design standpoint, we wanted Asterisk to be self-sufficient on this new Incredible PBX platform. Stated another way, we didn’t want Asterisk to fail just because Apache or MySQL had system failures since neither of them is required for Asterisk to function reliably in the first place. It’s one thing for your GUI or MySQL database to be inoperable. It’s quite another when it also brings down your entire phone system.

In summary, we are lifelong believers in the KISS principle. Keep It Simple, Stupid. As much as we love FreePBX, its system design is anything but simple. Configuration information is embedded in hundreds of HTML files, Linux templates, Asterisk configuration files including AstDB plus 100+ MySQL tables. By contrast, Asterisk-GUI uses a tiny collection of native Asterisk .conf files to configure virtually all its settings. We wanted to preserve that “pure Asterisk” simplicity.

One of the other real advantages of the Asterisk-GUI design is you can create something in the GUI and then review the Asterisk-generated code in /etc/asterisk to see exactly how the original Asterisk developers intended the feature to work. In addition to the learning experience, it makes it easy to debug coding errors and to make adjustments and customizations to meet individual needs without inadvertently bringing down the whole house of cards.

We wanted a product that was easy for an administrator to maintain, to update, AND to back up. After all, this is a phone system not a rocketship. It shouldn’t take a rocket scientist to maintain it. And it won’t.

Project Design 101: Preconfigured Trunks, Extensions & Routes

With these objectives in mind, we’ve made some design choices on the front end that are worth mentioning. Configuration settings for SIP, IAX, and Google Voice trunks give new users more headaches than any other single feature in a new PBX. So we’re taking much of the pain out of that process by providing 9 preconfigured trunks. Meet the Incredible 9: Google Voice, Vitelity, VoIP.ms, Les.net, IPcomms, DIDlogic, CallCentric, FutureNine, and Anveo Direct. Outbound calling is managed by routes that are tied to individual extensions. These can be adjusted quickly in the GUI. We’ve chosen to set up outbound calling for the Incredible 9 using preconfigured dialing prefixes. No prefix or a 1-prefix sends the call out through Google Voice and, if Google Voice isn’t available, then the call is routed through the next working outbound trunk in the order shown above. A prefix of 2-9 sends the call out through one of the preconfigured trunks. We’ve also included support for free worldwide iNum calling using either VoIP.ms or CallCentric. Both vendors will also provide you with a free iNum DID. Just dial your iNum prefix of 0 (CallCentric) or 90 (VoIP.ms) followed by the last 7 digits of any assigned iNum DID to place a free call. As usual, Lenny stands ready to provide 24/7 technical support through his iNum DID: And, of course, all of these settings can be modified or tweaked to your liking using Asterisk-GUI!

A word about the “Incredible 9″ providers. The major prerequisite for inclusion was communications compatibility with Asterisk without any firewall exposure of Asterisk ports. That means the provider had to support outbound and/or inbound calling without any port exposure of Asterisk to the Internet. Vitelity and Google have been major financial supporters of our projects over the years so they made the short list. Both also offer incredible pricing and feature-rich VoIP implementations. The others made the cut based upon great user satisfaction reports, free services of one type or another, or dirt cheap pricing. Can you add additional providers using Asterisk-GUI? Absolutely. But the “Incredible 9″ each can be activated in under 10 seconds after you’ve signed up for an account with your choice of providers. In the VoIP world, there’s little reason not to choose several since you only pay for the services you actually use, and we would encourage you to do so.

Incoming call processing also is preconfigured with some extensions, a ring group, a Stealth AutoAttendant, DISA, and an IVR with an assortment of Incredible PBX applications for Asterisk. All can be modified or embellished to meet your own requirements.

Bottom Line: You get a turnkey PBX that’s ready to go. It’s also easily configurable to meet your most demanding requirements. Incredible PBX delivers The Best of Both Worlds using native Asterisk code.

A Fresh Look at Managing Credentials

One of the more exasperating realities of password management with FreePBX is the number of places you have to look to find or change passwords. Some are stored in various Asterisk .conf files. Voicemail passwords are hidden away in text strings in voicemail.conf. Others are stored in MySQL tables. Some are encrypted, and some aren’t. Asterisk-GUI took a different approach and stores all passwords in the Asterisk .conf files in /etc/asterisk.

As talented as the FreePBX and Asterisk-GUI programmers are, we don’t trust any web-based application to remain secure if it’s directly exposed to the Internet. If you do, you’re either nuts or have plenty of money to burn. GUIs should be reserved for administrator use behind a secure firewall, period. In our new design, you need firewall whitelist privileges plus root or asterisk user privileges plus GUI admin user access to gain access to passwords. If all of these layers are compromised, passwords are the least of your worries.

We’ve taken password management one step further. As best we can given the design choices in Asterisk 11 and Asterisk-GUI, we’ve aggregated as many passwords as possible into new credentials config files: credentials-sip.conf, credentials-googlevoice.conf, and credentials-extensions.conf. There’s one for the “Incredible 9″ SIP providers. There’s one for Google Voice. And there’s a catchall for various passwords, PINs, and predefined CallerID numbers for various trunks. These are straight-forward text files that can be quickly edited using any text editor. Plug in your account names, passwords, and PINs. Optionally, adjust the providers’ server addresses as required. And you’re done. If you can tie your shoes, you can do this. Quick and functional, not fancy!

Redesigned Conferencing Solution for Asterisk 11

MeetMe conferencing as originally implemented in Asterisk-GUI required an external timing source. This timing source was provided by analog boards on some of the commercial hardware platforms on which Asterisk-GUI was deployed. For our target audience, we’re assuming that most people probably want to ditch Ma Bell and costly landlines as part of the migration to a new PBX platform. So, even though Asterisk-GUI still supports analog trunks, we have chosen to offer the Asterisk 11 Conference Bridge option which does not require an external timing source. The new Conference Bridge is preconfigured out of the box. Set up user and admin PINs. And you’re done. Dial C-O-N-F (3663) to join the conference.

The Baker’s Dozen Incredible PBX Apps: Alive and Well

We closed out Week One with some minor tweaking of several of our favorite Incredible PBX applications to accommodate the new Asterisk-GUI platform. We’re pleased to report that everything still works. Because of changes imposed by Google, you’ll need to jump through a few hoops to implement Speech Recognition support on this new Asterisk platform. All of the necessary software has already been put in place so all you need is an API key from Google. Once you obtain it, simply plug it into line 70 of speech-recog.agi. No other configuration is required. The affected applications are marked with an asterisk (*) below. But the good news is, if you’ve used these Nerd Vittles applications in the past, you’ll feel right at home.

Stay tuned for more and… HAPPY THANKSGIVING!

Continue reading Chapter 2

Originally published: Monday, November 24, 2014



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

VoIP Navigation Guide: Getting Started with Asterisk and FreePBX


When you were just getting started with Asterisk® in the early days, you had two choices: hire a consultant to build you an Asterisk system or start with Asterisk@Home and learn it yourself. That was a disaster for many folks. Times have changed, and there are literally dozens of aggregations and platforms from which to choose. But the question we continue to hear is “What’s the best way to get started?” Today’s VoIP Navigation Guide will help you make the right choices.

Before we begin, you need to do a little head-scratching yourself. Sit down with a pencil and paper (or a computer if you must) and jot down answers to our Top 10 Preliminary Questions:

  1. Is this for home or office use?
  2. How many simultaneous calls?
  3. How many users on the system?
  4. Will there be remote or traveling users?
  5. Is this a mission-critical system for you/others?
  6. What type & speed Internet service? Wi-Fi only?
  7. What is the skillset of those supporting the system?
  8. Do you want to babysit hardware for your system?
  9. What’s your initial and monthly budget for the project?
  10. What should happen to calls if your house/office burns down?

Skillset Matters! Let’s start with the obvious. The technical skillset of you and any other people that will be managing your VoIP server are critically important. This isn’t the old days where you only had to monitor people making long distance calls from within your own house. Once you connect a VoIP server to the Internet, anybody and everybody around the world can take a shot at your server and run up huge phone bills on your nickel unless you know what you’re doing or unless you deploy a server on which access is locked down to just you and trusted users and service providers.

We preach (regularly) that firewalls are essential if you’re going to deploy a VoIP server. In the home or office environment, that means that, in addition to your VoIP server, you also need a hardware-based firewall/router with no mapped ports to the VoIP server, period. Any other setup and it’s just a matter of time until you’re hacked.

In the hosted or cloud environment, it means at the very least a software-based firewall on your VoIP server with all access restricted to a whitelist of trusted users and providers. Any other setup and it’s just a matter of time until you’re hacked.

If you’re not qualified to manage either a hardware or software firewall, then your VoIP choices are limited. None of the major aggregations including PBX in a Flash, the FreePBX® Distro, AsteriskNOW, and Elastix provide any firewall protection as installed. While Fail2Ban is included, it is basically a log scanner which searches for failed login attempts and blocks IP addresses that make excessive login attempts. The major problem with Fail2Ban is that it takes time to run and, if your server is attacked from powerful servers, that may not happen until thousands of hack attempts have been executed.

We have attempted to address this problem with this summer’s new releases of Incredible PBX. In these new releases, whitelist access is locked down as part of the installation process. You have a choice of platforms.

On Cloud-based servers and depending upon your installation skills, we recommend:

On self-managed servers, you typically install the Linux operating system and then run the Incredible PBX installer. On smaller devices, we handle that for you. We recommend the following setups with the caveat that the old adage still applies: “You get what you pay for!” All four of the small hardware offerings below support WiFi-only operation. Just add the recommended WiFi USB dongle. For the CuBox-i, it’s built in. The VirtualBox setup takes less than 10 minutes.

Sizing Your Platform. Appropriate server and Internet capacity obviously turns on most of the answers you wrote down in the preliminary questionnaire. If the system will be used by less than a handful of people, you’re probably safe with the cloud-based solutions we’ve identified or one of the four low-cost devices listed above. Keep in mind that you need roughly 100Kbps of Internet bandwidth for each simultaneous VoIP call. If you have existing POTS lines from Ma Bell, those don’t consume Internet bandwidth but do consume local network resources. POTS line integration also requires additional hardware for each line. For less than 5 POTS lines, the OBi110 is an excellent choice. You’ll find it advertised in the right column of Nerd Vittles for under $50.

For up to a couple dozen low-call-volume employees, the RentPBX Cloud offering is a terrific bargain. It includes the necessary bandwidth not only to make calls but also to connect your extensions. When you get above those numbers of users or with heavy call volume, scaling matters. You don’t want to purchase a server only to discover on Day Two that it can’t handle the call volume. Here’s where the PBX in a Flash Forum can be a tremendous help. Describe your environment using the Top 10 Checklist from above. One of our hundreds of experts will lend a hand in recommending what you need to get started. Better yet, hire one of the gurus to handle the setup for you. It’ll save you thousands of dollars in headaches and easily pay for itself in future savings.

The PBX in a Flash Alternative. We haven’t mentioned PBX in a Flash as a solution for those just beginning their VoIP adventure. The reason is simple. The firewall is not preconfigured on PBX in a Flash, and somebody has got to do it unless your server is sitting behind a rock-solid, hardware-based firewall. The beauty of PBX in a Flash is that it’s incredibly flexible. You can choose not only the version of Asterisk and FreePBX to install, but you also can compile Asterisk with any collection of features desired. Once you get your feet wet with Incredible PBX, it’s our VoIP tool of choice, but it takes some skills on your part to run it safely. A good place to begin is the Nerd Vittles Quickstart Guide for PBX in a Flash 3. Enjoy!

Originally published: Wednesday, September 17, 2014


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! Please have a look and post your support questions there. Our forum is extremely friendly and is supported by literally hundreds of Asterisk gurus.



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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

FMC: The Future of Telephony with Vitelity’s vMobile and Asterisk in the Cloud




If making phone calls from a web browser is what you’ve always longed for, then you’re in good company with Google and its future direction in the telephony space. Call us old fashioned but this strikes us as a solution in desperate need of a problem. What’s wrong with a Plain Old Telephone or a smartphone for making connections with friends and business associates? The real head scratcher is the fact that the WebRTC and Hangouts push demonstrates that the wizards at Google are seriously out of touch with the next generation. Will our 14-year-old daughter use Skype or Hangouts or FaceTime? Sure. About once a month to chat with Grandma or to interact with cousins scattered around the country, it’s a terrific option. And the same is true in the business community. When you need to collaborate with a half dozen colleagues, conferencing applications are invaluable. But to meet 95% of day in and day out business requirements, a telephone or smartphone is the clear device of choice. So join us today in celebrating the end of Google Voice XMPP service and the beginning of a new and even more exciting VoIP era… sans Google.


Of course, if it were up to the next generation, telephone calls might completely disappear in favor of text messaging, Snapchat, Instagram, and any other platform that includes recorded photos or videos. Note the subtle difference. Kids really are not interested in live video interaction. They find posed images that tell a story much more appealing. Why? Because recorded photos and videos let users present their best face, their movie star pose, and their expression of what they want others to perceive they’re really like. In short, live video is too much like real life. Our conclusion for those targeting the next generation is you’d better come up with something better and quite different than Skype, Hangouts, and FaceTime.

It’s Fixed-Mobile Convergence, Stupid!

Now let’s return to our primary focus for today, the current business community. Suffice it to say, there are a dwindling number of what we used to call “desk jobs” where an employee arrives at his or her desk at 9 a.m. and leaves at 5 p.m. As more and more jobs are headed off shore, the telephone and smartphone have replaced the corporate desk as the most indispensable corporate fixture. Particularly in the American marketplace, what we see with most businesses is a management layer and an (upwardly) mobile force of salespeople, consultants, and implementers that interact primarily through PBXs in an office headquarters or home office together with smartphones for those that generally are on the road. Many of these Road Warriors don’t even have a home phone any longer.


The telephony Holy Grail for this new business model is Fixed-Mobile Convergence (FMC). It’s the ability to transparently move from place to place while retaining your corporate identity. Every employee from the night watchman in Miami to the salesperson making calls from a Starbucks in California to the CEO in New York has an extension on a PBX in the cloud together with the ability to accept and place calls using the company’s CallerID name and number, transfer calls, and participate in conference calls regardless of whether the phone instrument happens to be a desktop phone or a smartphone. Is this even possible? Well, as of last week, the answer is ABSOLUTELY.

Vitelity has been a long-time corporate sponsor of both the Nerd Vittles and PBX in a Flash open source projects so we were thrilled when we were offered a free, Samsung Galaxy S III to try out the new (live) vMobile service that took Best in Show honors at ITEXPO Miami in January. As Vitelity’s Chris Brown would probably tell you, it’s one thing to demonstrate a new technology at a trade show and quite another to bring it into production. But Vitelity did it:

What we want to stress up front is that we’ve received no special treatment in getting this to work. We received the phone, opened a support ticket to register the phone on Vitelity’s vMobile network, and plugged our new credentials into the phone so that it could be integrated into our PBX in a Flash server. Once the smartphone became an extension on our PBX, we could place calls through our PBX with the S3 using both WiFi and Sprint 3G/4G service. Switching between WiFi and cellular is totally transparent. The CallerID for all outbound calls was our standard PBX CallerID. We also could place calls to other extensions on the PBX by dialing a 4-digit extension while connected to WiFi or the Sprint network virtually anywhere. If you have 3-digit extensions, those are a problem over the Sprint network but we’ll show you a little trick to get them working as well.

Keep in mind that every call from the S3 goes out through the PBX just as if you were using a standard desktop phone as a hardwired extension. And it really doesn’t matter whether the S3 has a WiFi connection or a pure cellular connection on Sprint’s network. You receive calls on the S3 in much the same way. It’s just another extension on your PBX. If you want to add it to a ring group to process incoming calls, that works. If other users on your PBX wish to call the S3 directly using the extension number, that works as well. If you want to transfer a call, pressing ## on the S3 initiates the transfer just as if you were using a phone on your desk. When we say transparent convergence, we really do mean transparent. No recipient of a call from the vMobile S3 would have any idea whether you were sitting at a desk in the corporate headquarters in New York or in a seat on a Delta jet after landing in San Francisco. Both the call quality and the corporate CallerID would be identical. And your secretary on maternity leave at Grandma’s house still could reach you using her vMobile S3 by simply dialing your corporate extension.

So that’s the Fortune 500 view of the new VoIP universe. How about the little guy with a $15 a month PBX in a Flash server in the RentPBX cloud1, a couple mobile sales people, and a handful of construction workers that build swimming pools for a living? It works identically. Each has an S3 connected as an extension on the PIAF cloud server. And calls can be managed in exactly the same way they would be handled if everyone were sitting side-by-side at desks in an office headquarters somewhere. The silver lining of cloud computing is that it serves as the Great Equalizer between SOHO businesses and Fortune 500 companies. Asterisk® paired with inexpensive cloud hosting services such as RentPBX lets you mimic the Big Boys for pennies on the dollar. We think Vitelity has hit a bases loaded, home run with vMobile.


vMobile Pricing

We know what you’re thinking. “Since you got yours for free, what does it really cost??” The Galaxy S3 (or S4) is proprietary running Trebuchet 1.0, a (rooted) CyanogenMod version of Android’s KitKat. You can purchase these devices directly from the Vitelity Store. Currently, you can’t bring your own device. The refurbished S3 is $189 including warranty. Works perfectly! That’s what we’re using. Next, you’ll need a vMobile account for each phone. Unless you’re a Nerd Vittles reader, it’s $9.95 per month. That gets you free WiFi calling and data usage anywhere you can find an available WiFi hotspot. And text messaging is free. For calls and data using Sprint’s nationwide network, the calls are 2¢ a minute and the data is 2¢ per megabyte ($20 per gigabyte). For us, a typical day of data usage with an email account and light web use costs about a quarter. YMMV! So long as you configure Android to download application updates when connected to WiFi, data usage should not be a problem unless you’re into photos and streaming video. Android includes excellent tools for monitoring and even curbing your data usage if this is a concern.

vMobile Gotchas

Before we walk you through the setup process, let’s cover the gotchas. The list is short. First, we don’t recommend connecting vMobile devices to a PBX sitting behind a NAT-based firewall, or you may end up with some calls missing audio. The reason is NAT and quirky residential routers. If you think about it, when your S3 is inside the firewall and connected to WiFi, it will have an IP address on your private LAN just like your Asterisk server. When your S3 is outside your firewall on either a cellular connection or someone else’s WiFi network, it will have an IP address that is not on your private LAN. Others may be smarter than we are, but we couldn’t figure a way to have connections work reliably in both scenarios using most residential routers. You can configure your S3’s PBX extension for NAT=No or NAT=yes, but you can’t tell Asterisk how to change it depending upon where you are. One simple solution is to deploy these phones with a VPN connection to your Asterisk server sitting behind a NAT-based firewall. The more reliable solution is to build your PBX in a Flash server in the cloud with no NAT-based firewall. Then use an IPtables WhiteList (aka Travelin’ Man 3) to protect your server. From there, you can either interconnect the cloud-based server with a second PBX behind your firewall, or you can dispense with the local PBX entirely. Either way will eliminate the NAT issues with missing audio. In both cases, use NAT=yes for the vMobile extension.

Another wrinkle involves text messaging. Traditional text messages work fine; however, MMS still is problematic unless you initiate the outbound MMS session with the other recipient. It’s probably worth noting that Google Voice never got MMS working at all despite years of promises. This wasn’t a deal breaker for us, but it’s a bug that still is being worked on.

Finally, there’s Sprint. You either love ‘em or hate ‘em. We really haven’t used Sprint service in about eight years. In the Charleston area, the barely 3G service still is just as lousy as it was eight years ago. But, if you live in an area with good Sprint coverage and performance, this shouldn’t be an issue for you. And vMobile works fine in Charleston. You just won’t be surfing the web very often unless you have hours to kill… waiting. Additionally, dialing numbers with less than 4 numbers is a non-starter with Sprint, but we’ll show you a simple workaround to reach 3-digit local extensions from your vMobile device below.

With a service as revolutionary as vMobile with Sprint’s new FMC architecture, we can’t help thinking there may be other cellular carriers with an interest in deploying this technology sooner rather than later. But, given the vMobile feature set, Sprint is good enough for now especially when WiFi connectivity is available almost everywhere.




vMobile Configuration at Vitelity

For the Vitelity side of the setup, you first configure your smartphone using the (included) My Phone app. When the application is run, your cellphone number will be shown. Tapping the display about a dozen times will cause the phone’s setup to be reconfigured. Vitelity will provide you the secret key to activate your account. Next, you’ll log into the Vitelity portal and choose vMobile -> My Devices under My Products and Services. The account for your vMobile device will already exist. Clicking on the pull-down menu beside your vMobile device will let you create your SIP account on Vitelity’s server. Enter the IP address or FQDN of your Asterisk server and set up a very secure password. Your username will be the 10-digit phone number assigned to your vMobile phone. Save your settings and then choose the Edit option to view your setup. The portal will display your Username, Password, and FreePBX/Asterisk Connect Host name. Write them down for use when you configure your new extension using FreePBX®.




vMobile Configuration for Asterisk and PBX in a Flash

On the PBX in a Flash server, use a browser to open FreePBX. Choose Applications -> Extensions and add a new generic SIP device. For Display Name and User Extension, enter the 10-digit phone number assigned to your vMobile device. Under Secret, enter the password you assigned in Vitelity’s vMobile portal. Click Submit and reload FreePBX when prompted. Then edit the extension you just created. Set NAT=yes and change the Host entry from dynamic to the FQDN entry that was shown in Vitelity’s vMobile portal, e.g. 7209876542.mobilet103.sipclient.org. Update your configuration and restart FreePBX once again. Finally, from the Linux command prompt, restart Asterisk: amportal restart. If you’re using a WhiteList with IPtables such as Travelin’ Man 3, be sure to add a new WhiteList entry for your vMobile Host entry. Finally, add your vMobile extension to any desired Inbound Routes to make certain your vMobile device rings when desired.

You now should be able to place and receive calls on your vMobile device. If you want to be able to call 3-digit Asterisk extensions on both WiFi and while roaming on the Sprint cellular network, then you’ll need to add a little dialplan code since Sprint reserves 3-digit numbers for emergency services and will reject other calls with numbers of less than 4 digits. Here’s the simple fix. Always dial 3-digit extensions with a leading 0, e.g. 0701 to reach extension 701. We’ll strip off the leading zero before routing the call. The dialplan code below works whether you’re calling a local 3-digit extension or a 3-digit extension on an interconnected remote Asterisk server. Simply edit extensions_custom.conf in /etc/asterisk and insert the following code at the top of the [from-internal-custom] context. Then restart Asterisk: amportal restart. Note that we’ve set this up so that, if you have an extension 701 on both the local server and a remote server, the call will be connected to the local 701 extension. If you have different extension prefixes for different branch offices (e.g. 7XX in Atlanta and 8XX in Dallas), then this dialplan code will route the calls properly assuming you’ve configured an outbound route with the appropriate dial pattern for each branch office.

exten => _0XXX,1,Answer
exten => _0XXX,n,Wait(1)
exten => _0XXX,n,Set(NUM2CALL=${CALLERID(dnid):1})
exten => _0XXX,n,Dial(sip/${NUM2CALL})
exten => _0XXX,n,Dial(local/${NUM2CALL}@from-internal)
exten => _0XXX,n,Hangup

Vitelity vMobile Special for Nerd Vittles Readers

Now for the icing on the cake… We asked Vitelity if they would consider offering special pricing to Nerd Vittles readers and PBX in a Flash users. We’re pleased to report that Vitelity agreed. By using this special link when you sign up, the vMobile monthly fee will be $8.99 instead of $9.95. In addition, your first month is free with no activation fee. We told you last week that there was a very good reason for choosing Vitelity as your SIP provider. Now you know why.

And, if you’re new to Cloud Computing, take advantage of the RentPBX special for Nerd Vittles readers. $15 a month gets you your very own PBX in a Flash server in the Cloud. Just use this coupon code: PIAF2012. Enjoy!

Originally published: Thursday, May 15, 2014





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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

  1. RentPBX also is a corporate sponsor of the Nerd Vittles and PBX in a Flash projects. []

4 Months in Paradise: The Return of Free International VoIP Calling

With the impending implosion of Google Voice, it seemed appropriate to begin our quest for alternative termination providers. One of the real beauties of VoIP technology is you don’t have to put all of your eggs in one basket particularly in the termination department. It costs almost nothing to set up accounts with multiple providers for outbound calling. In addition to redundancy, the other clear advantage in using multiple providers for outbound calls is that you can take advantage of special rates to different destinations. So here’s the bargain of the week. If you have loved ones traveling to South America, Europe or Asia this summer, now’s your chance to sign up for VoIP service with FreeVoipDeal and enjoy four months of free calling to more than 50 countries around the world for every $15 of credits you purchase on their web site. Please note the fine print: “FreeVoipDeal reserves the right after a certain amount of calls to start charging the default rate.” There is no mention of what that “certain amount” happens to be. When your free calling finally ends, you can either purchase $15 of additional credits for 120 more “free” days or continue to call all of the previously free destinations for about 2¢ a minute.

The company behind FreeVoIPDeal is betamax which hosts over 30 sites offering varying deals to different countries. BEWARE: The prices change regularly. So a country that’s free today may suddenly cost money tomorrow. How does a mere mortal keep track? Well, betamax probably hopes that you won’t. But an enterprising individual named Robert Siemer has done the work for you. His backsla.sh/betamax web site automatically updates the pricing for all betamax sites every day! If this sounds like a lot of work to save a few cents a minute, you’d be right. And Vitelity which sponsors both the Nerd Vittles and PBX in a Flash projects offers consistently low rates to all of these countries. You’ll find a DID special at the end of this article, and their excellent international rate table is available at this link.

Setting Up an Account. Before you can set up a trunk in PBX in a Flash, you’ll first need to create a FreeVoipDeal account. In the “old days” this required use of their Windows client to obtain your credentials. Now you can simply create an account on the web site at this link. You’ll need either a regular land line or a cell phone number to verify your registration. Once you’re set up and you’ve deposited at least 10 euros (about $15) in your account, it’s time to set up a SIP trunk and outbound route in PBX in a Flash.

Configuring a Trunk with PBX in a Flash. Assuming you already have a phone registered to an extension in PBX in a Flash, it’s a one-minute drill to configure a trunk and outbound route to support FreeVoipDeal. Using a browser, log into FreePBX® using your maint username and password. Choose Connectivity -> Trunks -> Add SIP Trunk. Name the trunk: FreeVoipDeal. For the Dialed Number Manipulation Rules, enter Prepend: 1 and Match Pattern: NXXNXXXXXX. Clear out all of the default entries in Outgoing and Incoming Settings. Then, in Outgoing Settings, enter Trunk Name: freevoipdeal. For the PEER Details, enter the following using your actual account USERNAME and PASSWORD. Then SAVE your settings and reload FreePBX.

username=USERNAME
authuser=USERNAME
secret=PASSWORD
type=peer
qualify=yes
nat=yes
insecure=port,invite
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
dtmfmode=auto
disallow=all
canreinvite=no
allow=ulaw

There’s no need to enter a CallerID number. All of the outgoing calls will be delivered as ANONYMOUS. You also won’t need to register with the provider since Asterisk® can handle this on the fly using your credentials entered above.

Configuring an Outbound Route with PBX in a Flash. One more step, and you’ll be ready to start making calls. Choose Connectivity -> Outbound Routes. For the Route Name, enter: FreeVoIPDeal. For the Dial Pattern to make U.S. calls, enter: NXXNXXXXXX. If you want to force callers to dial a prefix to use the FreeVoipDeal trunk, then enter a 9 or some other number in the Prefix field. For Trunk Sequence 0, choose: FreeVoipDeal. Click Submit Changes and restart FreePBX when prompted. You’re done!

Making Your First Call. Using a phone or softphone logged into your server, dial the prefix (if any) plus the 10-digit number of someone in the United States. When the called party answers, make sure you can hear the called party and vice versa. If not, open Settings -> SIP Settings in FreePBX and add your External IP and Local Network settings. Also make certain the NAT entry is set to YES.

Configuring Your Server for International Calls. We do not recommend configuring your server to permit international calls to everywhere. The reason is simple. If strangers manage to access one of your extensions, they can run up your phone bill in a hurry. For this reason, we also strongly recommend that you do not configure automatic credit card replenishment with any VoIP provider!

For international calling, we recommend you add a separate Dial Pattern to both your FreeVoipDeal trunk AND the outbound route for each country code you wish to enable. Here is the complete list of codes. For example, to allow calls to Germany from another country, you’d add 49XXXXXXXXXX, save your changes, and reload FreePBX.

Spoofing Your CallerID. If you first verify that you own a number by using the web portal, you then can spoof the outbound CallerID using the number you verified. Just add the following entries to your trunk settings replacing 9991234567 with your verified CallerID number. Special thanks to @hillclimber on the PIAF Forum for the tip.
fromuser=0019991234567
sendrpid=yes

Originally published: Friday, April 25, 2014




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


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 


Some Recent Nerd Vittles Articles of Interest…

Obivoice = OBi Heaven: Dumping Google Voice for Less Than 10¢ a Day

What a difference a week makes! When we wrote last week’s article about netTALK and their terrific pricing, we were pleased to report that at least one company could offer a drop-in replacement for Google Voice without breaking the bank. But, alas, all is not well in netTALK Land. For openers, the Better Business Bureau revoked their accreditation last June because of failure to respond to or resolve technical complaints. And a recent SEC Filing paints a fairly bleak picture of the company’s financial condition. Special thanks to Gershom1624 for his sleuthing efforts. This merely reinforces the difficulty of providing reliable, unlimited VoIP service at the $2.50 a month price point. But we firmly believe $2.50 is the magic price point, and it is achievable with some safeguards for the provider, i.e. residential service, no call centers, no 10,000 minutes-a-month customers. My mom loved the telephone, but she never spent 5 hours a day on the telephone. There also has to be some tradeoff in the level of support customers can expect. If customers tie up expensive support reps with multiple calls, the pricing matrix falls apart very quickly. And that brings us to this week.


Let’s review the Wish List for those that missed last week’s article. We want a drop-in replacement for Google Voice on both the OBi110 (stand-alone with any POTS telephone) and Asterisk® (PBX) platforms. It needs to provide unlimited (within reason) calling in the U.S. and Canada. It needs a feature set that is fairly comparable to Google Voice. It needs to include E911 service because the federal government says so. We don’t care much about support as long as the setup process is well-documented, the service is reliable, and calls sound great. Charging for support requests to resolve issues that aren’t the company’s fault is perfectly fine with us. But the price point for unlimited calling needs to be $2.50 a month, i.e. $30 a year or $60 every two years for the math-challenged. We’d prefer no tips, taxes, or fees. We want to keep our existing number. And, lest we forget, the company must promise to stay in business and never raise prices… forever.

Suppose we could find you a company that, with a 2-year commitment, could provide all of the above (minus the last sentence) plus fax support including a web page to send outgoing faxes from attachments, free calling and a mobile app for your iOS and Android devices, Visual Voicemail with voicemail transcription as well as email delivery of voicemail messages, call forwarding, call waiting, CallerID spoofing for any number you own, and unbelievable customer service. Not sure about the service? How about a 30-day free trial with 60 free minutes?

Let us introduce you to Obivoice. Don’t be alarmed by the one-year price of $40. The two-year price is just $60. But it doesn’t cost you a nickel to sign up and try the service. Obivoice is a pure SIP provider so the setup with PBX in a Flash™ or an OBi110™ takes only a couple minutes. Here’s the SIP trunk setup for PBX in a Flash using FreePBX®. All you need is your SIP credentials and phone number once you’ve signed up for an account. Plug in your 10-digit phone number in the Outbound CallerID and Register String, replace 1234 with your Account Number in the username, fromuser, and Register String, and replace yourpassword with your real Password in the secret and Register String.

Next, build yourself an Inbound Route with your 10-digit DID and point it to your favorite PBX destination. Finally, create an Outbound Route using obivoice as the Trunk Sequence, and you’re all set. It doesn’t get any easier than that.

We don’t think you will but, if you need assistance setting this up, head over to the PIAF Forum where there’s a lively discussion about Obivoice already.

The OBi110 setup is just as easy. Plug in sms.intelafone.com as the ProxyServer and OutboundProxy in your ITSP Profile, add your SIP credentials in the SP1 Voice Services dialog, and forward (or transfer) your existing Google Voice number to Obivoice. Done! Obivoice’s complete tutorial is available here.

Let us close with our own customer service story. We were so excited about this new service when it was announced yesterday that we actually clicked the wrong button and signed up for the wrong plan. Of course, it only takes a minute to get that sinking feeling in your stomach when you know you’ve screwed up. So late yesterday (Sunday night!) I opened a support ticket and asked to either cancel the wrong plan so that I could reenlist or to transfer to the $60 two-year plan. At 1:30 a.m. this morning, I got an email back from customer service indicating that the plan had been adjusted and that I had been billed for the price difference. WOW!

Run, don’t walk, to sign up for Obivoice. It’s that great!

p.s. The Obivoice jingle in their YouTube video is as good as their calls. We want it for our Music on Hold!

Originally published: Monday, January 13, 2014




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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for all of us.


 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity. 


Some Recent Nerd Vittles Articles of Interest…

The Definitive VoIP Quick Start Guide: Introducing PBX in a Flash 2.0.6.5.0

What a difference a year makes in the VoIP World! We now have a rock-solid, reliable Asterisk® 11 release and an equally stable FreePBX 2.11 on which to build state-of-the-art VoIP servers. If you’re new to the VoIP community, watch this video before you proceed.


Now let us welcome you to the World of PBX in a Flash™. This is our best release ever whether you’re a total newbie or an experienced Asterisk developer. You can’t really appreciate what goes into an open source product like PBX in a Flash until you try doing it yourself. The sad part is we and the CentOS™ development team are part of a dwindling few non-commercial entities that still are in the open source “business.” If you want to actually learn about Asterisk from the ground up using pure source code to customize your VoIP deployment, PBX in a Flash has no competition because your only other option is to roll your own starting with a Linux DVD. So our extra special kudos go to Tom King, who once again has produced a real masterpiece in that it is very simple for a first-time user to deploy and, at the same time, incredibly flexible for the most experienced Asterisk developer. The new PIAF 2.0.6.5.0 ISOs not only provide a choice of Asterisk® and FreePBX® versions to get you started. But now you can build and deploy standalone servers for SugarCRM™, NeoRouter™ VPN, YATE™, FreeSwitch™, and OpenFire™ XMPP using the 32-bit and 64-bit PIAF™ ISOs. So let’s get started.

Making a Hardware Selection

We’re going to assume that you need a VoIP telephony solution that will support an office of up to several dozen employees and that you have an Internet connection that will support whatever your simultaneous call volume happens to be. This is above and beyond your normal Internet traffic. To keep it simple, you need 100Kbps of bandwidth in both directions for each call.1 And you need a router/firewall that can prioritize VoIP traffic so that all your employees playing Angry Birds won’t cause degradation in VoIP call quality. Almost any good home router can now provide this functionality. Remember to disable ALG on your router, and it’s smooth sailing.

For computer hardware, you’ll need a dedicated machine. There are many good choices. Unless you have a burning desire to preserve your ties with Ma Bell, we recommend limiting your Ma Bell lines to your main number. Most phone companies can provide a service called multi-channel forwarding that lets multiple inbound calls to your main number be routed to one or more VoIP DIDs much like companies do with 800-number calls. If this works for you, then any good dual-core Atom computer will suffice. You’ll find lots of suggestions in this thread. And the prices generally are in the $200-$400 range. For larger companies and to increase Asterisk’s capacity with beefier hardware, see these stress test results.

If your requirements involve retention of dozens of Ma Bell lines and complex routing of calls to multiple offices, then we would strongly recommend you spend a couple thousand dollars with one of our consultants. They’re the best in the business, and they do this for a living. They can easily save you the cost of their services by guiding you through the hardware selection process. They also have turnkey phone systems using much the same technology as you’ll find in PBX in a Flash. You won’t hurt our feelings. :-)

Choosing the Right PIAF Platform

We get asked this question about a hundred times a week on the forums so here goes. There are more than two dozen permutations and combinations of CentOS, Asterisk, and FreePBX to choose from when you decide to deploy PBX in a Flash. We always recommend the latest version of CentOS because it tends to be the most stable and also supports the most new hardware. You have a choice to make between a 32-bit OS or 64-bit. Our preference is the 32-bit platform because it is better supported. The performance difference is virtually unnoticeable for most VoIP applications. With Asterisk, we always recommend an LTS release because these have long-term support. That narrows your choices to Asterisk 1.8 or Asterisk 11. At this juncture, we think you’d be crazy to install anything other than Asterisk 11. It’s incredibly reliable and stable, and it will be supported for years to come. It also supports Digium Phones. The bottom line is that Asterisk 11 is the latest and greatest with the best feature set. If we were building a system for a commercial business, it would be our hands-down choice. In the PBX in a Flash world, we have colors for various versions of PBX in a Flash that support different versions of Asterisk. Asterisk 11.6 happens to be the latest PIAF-Green, and we recommend you install it with the latest version of FreePBX as well, 2.11.0.11

Choosing the Right Phones

If there is one thing that will kill any new VoIP deployment, it’s choosing the wrong phones. If you value your career, you’ll let that be an organization-driven decision after carefully reviewing at least 6-12 phones that won’t cause you daily heartburn. You and your budget team can figure out the price points that work in your organization keeping in mind that not everyone needs the same type of telephone. Depending upon your staffing, the issue becomes how many different phone sets are you and your colleagues capable of supporting and maintaining on a long term basis.

Schmooze Com has released their commercial End Point Manager (EPM) at a price point of $39 per server. They’ve been using the application internally to support their commercial customers for over a year. Suffice it to say, it’s the best money you will ever spend. You can sign up for an account with Schmooze through our commercial support site and purchase the software now. You can review the Admin User Guide here. The beauty of this software is it gives you the flexibility to support literally hundreds of different VoIP phones and devices almost effortlessly. Using a browser, you can configure and reconfigure almost any VoIP phone or device on the market in a matter of minutes. So the question becomes which phones should you show your business associates. That again should be a decision by you and your management and budget teams, but collect some information from end-users first. Choose a half dozen representative users in your company and get each of them to fill out a questionnaire documenting their 10 most frequent daily phone calls and listing each step of how they processed those calls. That will give you a good idea about types and variety of phones you need to consider for different groups of users. Cheaper rarely is better. Keep in mind that phones can last a very long time, even lousy ones. So choose carefully.

The phone brands that we would seriously consider include Yealink, Aastra, Snom, Digium, Mitel, Polycom, Cisco, and Grandstream. Do you need BLF, call parking or multiple line buttons, a hold button, conferencing, speakerphone, HD voice, power over Ethernet support, distinctive ringtones for internal and various types of external calls, Bluetooth, WiFi, web, SMS, or email access, an extra network port for a computer, headset support, customizable buttons (how many?), quick dial keys, custom software, XML provisioning, VPN support? How easy is it to transfer a call? Do you need to mimic key telephones? Also consider color screens, touch screens, busy lamp indicators, extension modules (what capacity?). What do we personally use: Yealink’s T46G is our favorite, and we also have several Digium phones of various types, a couple of Aastra phones, a Grandstream GXP2200, and a collection of Panasonic cordless DECT phones, a fax machine as well as a Samsung Galaxy S4 and Moto X connected through an OBi202 with an OBiBT Bluetooth Adapter.

Installing PBX in a Flash

With the office politics out of the way, let’s get to the fun stuff.

For most deployments, choose the default install by pressing Enter.

Leave the UTC System Clock option unchecked and pick your Time Zone. Tab to OK and press Enter.

Choose a very secure Root Password. Tab to OK and press Enter. Your server will whir away for 5-10 minutes installing CentOS 6.4. When the reboot begins, remove the DVD or USB thumb drive.

Log into your server as root from either the console or an SSH connection to the IP address displayed on your server. Unless you need to install custom hardware drivers, choose the first option to install PBX in a Flash.

For today, we’re installing PBX in a Flash. So leave it highlighted, tab to OK, and press Enter.

Now pick your PIAF flavor, tab to OK, and press Enter. You’ll note there are some new colors. :-)

The PIAF Configuration Wizard will load. Press Enter to begin.

Unlike any other aggregation, PIAF gives you the opportunity to fully configure Asterisk using make menuconfig if you know what you’re doing. For everyone else, type N and then confirm your choice.

Next, you’ll need to choose your Time Zone again for PHP and FreePBX. Don’t worry if yours is missing. A new timezone-setup utility is also to reconfigure this to any worldwide time zone once the install has completed.

Next, choose your version of FreePBX to install. If you plan to also install Incredible PBX and Incredible Fax:

Incredible PBX 3 requires PIAF-Purple and FreePBX 2.9
Incredible PBX 4 requires PIAF-Purple and FreePBX 2.10 (32-bit only)
Incredible PBX 11 requires PIAF-Green and FreePBX 2.11 (recommended!)

Finally, you need to choose a very secure maint password for access to FreePBX using a browser. You can pick your own, or the installer will generate one for you. Don’t forget it.

The installer will give you one last chance to make changes. If everything looks correct, press the Enter key and go have lunch. Be sure you have a working Internet connection to your server before you leave. :wink:

In about 30-60 minutes, your server will reboot. You should be able to log in as root again using your root password. Write down the IP address of your server from the status display (above) and verify that everything installed properly. Note that Samba is disabled by default. If you want to use your server with Windows Networking, run configure-samba once your server is up and running and you’ve logged in. You also can ignore the MySQL DOWN alert shown above. Yours won’t say that. We’ve been experimenting with MariaDB as a MySQL replacement. You can read all about it in the Developers’ Corner of the PIAF Forum.

Configuring PBX in a Flash

Most PIAF Configuration is accomplished using the FreePBX Web GUI. Point your browser to the IP address shown in the status display above to display your PIAF Home Page. Click on the Users tab. Click FreePBX Administration. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in the Config Module phase of the PBX in a Flash installation procedure above.

If you’re new to Asterisk and FreePBX, here’s the one paragraph primer on what needs to happen before you can make free calls with Google Voice. You’ll obviously need a free Google Voice account. This gets you a phone number for people to call you and a vehicle to place calls to plain old telephones throughout the U.S. and Canada at no cost. You’ll also need a softphone or SIP phone to actually place and receive calls. YATE makes a free softphone for PCs, Macs, and Linux machines so download your favorite and install it on your desktop. Phones connect to extensions in FreePBX to work with PBX in a Flash. Extensions talk to trunks (like Google Voice) to make and receive calls. FreePBX uses outbound routes to direct outgoing calls from extensions to trunks, and FreePBX uses inbound routes to route incoming calls from trunks to extensions to make your phones ring. In a nutshell, that’s how a PBX works. There are lots of bells and whistles that you can explore down the road. FreePBX now has some of the best documentation in the business. Start here.

To get a minimal system functioning to make and receive calls, here’s the 2-minute drill. You’ll need to set up at least one extension with voicemail, and we’ll configure a free Google Voice account for free calls in the U.S. and Canada. Next, we’ll set up inbound and outbound routes to manage incoming and outgoing calls. Finally, we’ll add a phone with your extension credentials.

A Few Words About Security. PBX in a Flash has been engineered to run on a server sitting safely behind a hardware-based firewall with NO port exposure from the Internet. Leave it that way! It’s your wallet and phone bill that are at stake. If you’re running PBX in a Flash in a hosted environment with no hardware-based firewall, then immediately read and heed our setup instructions for Securing Your VoIP in the Cloud Server. We would encourage you to visit your PIAF Home Page regularly. It’s our primary way of alerting you to security issues which arise. You’ll see them posted (with links) in the RSS Feed shown above. If you prefer, you can subscribe to the PIAF RSS Feed or follow us on Twitter. For late-breaking enhancements, you also should regularly visit the Bug Reporting & Fixes Topic on the PIAF Forum.

Extension Setup. Now let’s set up an extension to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone’s GUI to add bells and whistles. In FreePBX 2.10 or 2.11, to create extension 201 (don’t start with 200), click Applications, Extensions, Generic SIP Device, Submit. Then fill in the following blanks USING VERY SECURE PASSWORDS and leaving the defaults in the other fields for the time being.

User Extension … 201
Display Name … Home
Outbound CID … [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID … [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]

Device Options
secret … 1299864Xyz [randomly generated]
dtmfmode … rfc2833
Voicemail Status … Enabled
voicemail password … 14332 [make this unique AND secure!]
email address … yourname@yourdomain.com [if you want voicemail messages emailed to you]
pager email address … yourname@yourdomain.com [if you want to be paged when voicemail messages arrive]
email attachment … yes [if you want the voicemail message included in email]
play CID … yes [if you want the CallerID played when you retrieve message]
play envelope … yes [if you want date/time of the message played before the message]
delete Vmail … yes [if you want the voicemail message deleted after it’s emailed to you]
vm options … callback=from-internal [to enable automatic callbacks by pressing 3,2 after playing a voicemail message]
vm context … default

Write down the passwords. You’ll need them to configure your SIP phone.

Extension Security. We cannot overstress the need to make your extension passwords secure. All the firewalls in the world won’t protect you from malicious phone calls on your nickel if you use your extension number or something like 1234 for your extension password if your SIP or IAX ports happen to be exposed to the Internet.

In addition to making up secure passwords, the latest versions of FreePBX also let you define the IP address or subnet that can access each of your extensions. Use it!!! Once the extensions are created, edit each one and modify the permit field to specify the actual IP address or subnet of each phone on your system. A specific IP address entry should look like this: 192.168.1.142/255.255.255.255. If most of your phones are on a private LAN, you may prefer to use a subnet entry in the permit field like this: 192.168.1.0/255.255.255.0 using your actual subnet.

Adding a Google Voice Trunk. There are lots of trunk providers, and one of the real beauties of having your own PBX is that you don’t have to put all of your eggs in the same basket… unlike the AT&T days. We would encourage you to take advantage of this flexibility. With most providers, you don’t pay anything except when you actually use their service so you have nothing to lose.

For today, we’re going to take advantage of Google’s current offer of free calling in the U.S. and Canada through the end of 2013. You also get a free phone number in your choice of area codes. PBX in a Flash now installs a Google Voice module under FreePBX -> Connectivity that lets you set up your Google Voice account with PBX in a Flash in just a few seconds once you have your credentials.

A Word to the Wise: All good things come to an end… especially those that are free. So plan ahead with some alternate providers that keep your phones working should Google decide to pull the plug or change the terms with Google Voice.

Signing Up for Google Voice. You’ll need a dedicated Google Voice account to support PBX in a Flash. 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!

We’ve tested this extensively using an existing Gmail account rather than creating a separate account. Take our word for it. Inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So… set up a dedicated Gmail and Google Voice account2, and use it exclusively with PBX in a Flash. Google Voice no longer is by invitation only. If you’re in the U.S. or have a friend that is, head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: 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 PBX in a Flash to function with Google Voice! Otherwise, inbound and/or outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings and enable it. Be sure to try one call each way from Google Chat in Gmail. Then disable Google Chat in GMail for this account. Otherwise, it won’t work with PIAF.

While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:

  • Call ScreeningOFF
  • Call PresentationOFF
  • Caller ID (In)Display Caller’s Number
  • Caller ID (Out)Don’t Change Anything
  • Do Not DisturbOFF
  • Call Options (Enable Recording)OFF
  • Global Spam FilteringON

Click Save Changes once you adjust 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.

Configuring Google Voice Trunk in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize PBX in a Flash to meet your specific needs. Click the Connectivity tab in FreePBX 2.11 and choose Google Voice [Motif]. To Add a new Google Voice account, just fill out the form. NOTE: The form has changed from prior releases of FreePBX. Do NOT check the last box: Send Unanswered to GoogeVoice Voicemail, or you may have problems receiving incoming calls.

Google Voice Username is your Google Voice account name without @gmail.com. Password is your Google Voice password. NOTE: Don’t use 2-stage password protection in this Google Voice account! Phone Number is your 10-digit Google Voice number. Next, check only the first two boxes: Add Trunk and Add Outbound Routes. Then click Submit Changes and reload FreePBX. Down the road, you can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill. For Google Apps support, see this post on the PIAF Forum.

Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. It also provides redundancy which costs you nothing if you don’t use the backup providers. The Google Voice module actually configures an Outbound Route for 10-digit Google Voice calling as part of the automatic setup. If this meets your requirements, then you can skip this step for today.

Inbound Routes. An Inbound Route tells PBX in a Flash how to route incoming calls. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we’ll build a simple route that directs your Google Voice calls to extension 201. Choose Connectivity -> Inbound Routes, leave all of the settings at their default values except enter your 10-digit Google Voice number in the DID Number field. Enable CallerID lookups by choosing CallerID Superfecta in the CID Lookup Source pulldown. Then move to the Set Destination section and choose Extensions in the left pull-down and 201 in the extension pull-down. Now click Submit and save your changes. That will assure that incoming Google Voice calls are routed to extension 201.

IMPORTANT: Before Google Voice calling will actually work, you must restart Asterisk from the Linux command line interface. Log into your server as root and issue this command: amportal restart.

Eliminating Audio and DTMF Problems. You can avoid one-way audio on calls and touchtones that don’t work with these simple settings in FreePBX: Settings -> Asterisk SIP Settings. Just plug in your public IP address and your private IP subnet. Then set ULAW as the only Audio Codec.

General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. In FreePBX 2.11, choose Admin -> Module Admin and click on the Upgrade Notifications shield on the right. Plug in your email address, click Submit, and save your changes. Done!

Setting Up a Desktop Softphone. PBX in a Flash supports all kinds of telephones, but we’ll start with the easy (free) one today. You can move on to “real phones” once you’re smitten with the VoIP bug. For today, you’ll need to download a softphone to your desktop PC or Mac.

The easiest way to get started is to set up a YATE softphone on your Desktop computer. Versions are available at no cost for Macs, PCs, and Linux machines. Just download the appropriate one and install it from this link. Once installed, it’s a simple matter to plug in your extension credentials and start making calls. Run the application and choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of your server, 201 for your account name, and whatever password you created for the extension. Click OK.

Once you are registered to extension 201, close the Account window. Then click on YATE’s Telephony Tab and place your first call. It’s that easy!

Monitoring Call Progress with Asterisk. That about covers the basics. We’ll leave you with a tip on how to monitor what’s happening with your PBX. There are several good tools within the FreePBX GUI. You’ll find them under the Reports tab. In addition, Asterisk has its own Command Line Interface (CLI) that is accessible from the Linux command prompt. Just execute the following command while logged in as root: asterisk -rvvvvvvvvvv.

What’s Next? We’ve barely scratched the surface of what you can do with PBX in a Flash. Log into your server as root and type help-pbx for a list of simple install scripts that can add almost any function you can imagine. And Incredible PBX 11 and Incredible Fax can be installed in under 2 minutes to provide you almost every Asterisk application on the planet. You can read the complete tutorial here. In addition, Travelin’ Man 3 can be installed as part of Incredible PBX for rock-solid Internet security. If you care about your wallet, add Travelin’ Man to your server!

New App of the Week. We’re pleased to introduce Trunk Failure Email Alerts for Asterisk supporting SIP, IAX2, and Google Motif trunks. Just insert your email address in this little script and run it every hour as a cron job. You’ll get an email alert whenever any of your VoIP trunks fail. Enjoy!

VoIP Experts on Twitter. GetVoip.com has just released their list of The Top 50 VoIP Experts to Follow on Twitter. It’s a great read… but we may be biased. :wink:

Join Google+ Today. For the latest VoIP and technology news, come follow us on Google+ and join CircleCount.com for a terrific overview of your Google+ friends and their hometowns.

Originally published: Tuesday, December 17, 2013




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