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JUST RELEASED: Visit the Incredible PBX WikiSurfing the Google Wave
We’ve spent a week getting to know Google Wave using Chrome along with 100,000 of our closest friends. We wanted to give you a status report. Hype aside, Google Wave is an incredible tool when used for the right purpose. If you’ve been asleep or hiding under a rock for the past two weeks and missed the party, here’s a quick summary of Google’s latest invention. It’s a bird, it’s a plane… actually it’s a collaboration and communications platform that brings the full richness of Web 2.0 to your desktop. Some have suggested that it’s what email would look like if it were invented today. Our discussion focuses on the web-based Wave client, but Google Wave also is an open source development toolkit, and we’ll get to that one day soon.
Much has been written about Google Wave’s capabilities, and we won’t repeat that here. Instead, we want to address Google Wave’s potential and what we see as some of the present shortcomings of the product. We fully appreciate that this is a preview, and many of our concerns may yet be addressed before Google Wave becomes available to the general public. We can’t help chuckling at the realization that, in less than 30 years, we’ve now come full circle in data processing. What began as mainframe computing evolved into personal computing. And now Google Wave brings us much closer to being back where we started except for a state-of-the-art user interface and a new name: Cloud Computing. If IBM had addressed the user interface issues with mainframe computing, they probably never would have lost their market in the first place.
The screenshot above really can’t do justice to the richness of the client interface because you truly need a monitor as wide as your desk to get the most out of Google Wave. No, you won’t be using this on your cellphone or PDA… at least not well. For openers, Google Wave provides real-time collaboration so you actually see folks typing into various message threads (called waves) in the Land of Google. When you create a new wave, you "invite" other people in your Contacts to the wave. This puts the wave at the top of their Inbox in bold-faced type, akin to what Gmail would do with an incoming email.
There’s another frame to the right of your Inbox which actually displays the complete wave that you have selected so you’re never really jumping back and forth between selecting waves and reading them. What takes a bit of getting used to is the fact that both your Inbox and the wave you are currently reading may be changing every second with input from literally dozens of your associates or strangers if the wave you’re reading happened to be designated as public.
There’s one other dramatic difference in waves and threads of email messages. Other folks can change your stuff. As a collaboration tool with close associates, this might be desirable. With public waves, it would be a nightmare in the real world. And we don’t consider the 100,000 Google Wave previewers the real world. They are for the most part well-behaved probably out of fear that they’d be booted out if they behaved badly. That isn’t the real world as we all know. And the current Google Wave design would let a single creep destroy virtually every public wave in minutes using bots and malicious changes to documents. As presently designed, there would be little recourse other than replaying what your wave used to look like. You really can’t put Humpty back together again as some have already discovered.
Richard Nixon learned the hard way that tape recorders can be a blessing and a curse. Much has been made of the capability Google Wave offers to replay a wave so that you can playback the development of a thread of messages and see who added or deleted what and when. Google has touted the fact that everything is preserved. Well, not quite. First, there’s no capability at least presently to scroll back to a certain place in the timeline and recreate a new wave up to there. The most you can retrieve is a single posting. Second, anybody with access to the wave can use this timeline feature so wave restoration wouldn’t necessarily be desirable unless it were restricted to the original author of the wave. And, third, at least at Google someone knows how to cheat the system and delete stuff from the timeline. We only discovered this in reviewing the first public porn wave which started out prim and proper enough but quickly gathered steam when someone posted a collection of NSFW (or anywhere else) photos from their favorite collection. Within a few minutes, the postings quietly disappeared. Being the careful reviewer that we are, we immediately reached for the Playback button to check the history of the wave. Sure enough, the raunchy photos were still there. But, by the next morning, they had completely vanished from the chronology. So much for the official stance that nothing ever disappears. The real disappointment with the replay function is the lack of any capability to restore an entire wave. Because only individual messages (known as blips) can be recovered, this would prove to be next to worthless in a complex wave with hundreds of postings.
That brings us to the issue of whether public waves really make sense given the world in which we live. The good news is it works much better than IRC because of the richness of the content with attachments and hyperlinks. But, at least for public waves, the ability to edit someone else’s stuff would have to go. We try not to focus on legal nightmares in reviewing new software, but one can’t help wondering what would happen if one were to post something complimentary about a neighbor in a public wave and then another neighbor altered your posting by falsely accusing the individual of sleeping around with half of the neighborhood. Obviously, there’s still a good bit of work to do on the security front and in deciding whether allowing others to amend someone else’s postings is a good idea. Whether Google gets the security piece right will ultimately determine the success of Google Wave.
For public waves, it’s a no-brainer. You just can’t! And, to be honest, in reviewing hundreds of public waves in the preview, we can’t recall a single instance where this functionality would have been necessary. In a true (private) collaborative project, it would be wonderful but color-coding of text or some other method of identifying who wrote what would be absolutely essential from both a practical and legal standpoint. Both Microsoft Word and WordPerfect have had this capability forever. The simple way in Google Wave would be to add user’s pictures with a colored border and matching colored text whenever they make changes to someone else’s posting. With this addition, Google Wave could become a wonderful collaborative tool in both legal and technical environments.
And, speaking of word processing, Google Wave falls a bit short on the word processing scale. Despite the richness of Google’s knol platform, some of that functionality still is not available in Google Wave. The text editing and formatting is much akin to what’s available in a typical email client. You can change fonts, adjust color, indent, add hyperlinks and images, but that’s about as far as it goes. There are no headers, footers, footnotes, etc. So you can’t easily transform a wave into a formatted document for printing at this juncture. But that may come as development continues.
There are a few other things still on our Wish List. First, we’d love for Google Wave to evolve into a tool that can replace today’s forums which are not much more functional than BBS software was two decades ago. Once there is administrator control of rollback and protection of waves by granular access rights to functions, bots, and gadgets as well as the ability to block users and ranges of IP addresses, this should happen. Second, we obviously want the ability to include either read-only or read-write access to waves in a blog or web site. We already have the web site functionality working (see below for a sample), but you currently need a Google Wave account to access it. Third, we really want to assimilate all of the tools we use into the Google Wave Desktop so that everything is accessible in one place. That’s what Cloud Computing is all about, and Google Wave comes closer than anything else in meeting that need. You already can access Gmail on your Google Wave desktop and any web site that can be framed can be included in a wave as an iFrame. That doesn’t leave much once the security feature set is in place to protect all the components.
Finally, we’ll close with a brief mention of the coolest feature of Google Wave. That is its expandability which is enabled by incorporating bots and gadgets into any wave. As you might imagine, these extensions can do almost anything… good or bad. Here’s a short list of what has been developed and what’s already on the radar in just a few short weeks:
Chatbots
Eliza – ogenex@appspot.com – An implementation of the Eliza chatbot borrowed from the NLTK.
Elize – elizarobot@appspot.com – Is one of the first robots that was created by non Googler and is very useful if you are feeling alone in your Google Wave client.
Rude chatbot – notatory@appspot.com – An obnoxious chatbot borrowed from the Natural Language Processing Toolkit.
TooAngel Wave – In Progress – tooangel-wave@appspot.com – A self learning robot, that will respond to a reply in a more humanoid wayConversion
BotURL – boturl@appspot.com – A URL Linker that replaces full URLs with hyperlinks.
Calcbot – calcbot@appspot.com – This bot will do in place calculations for simple mathematical expressions and allow you to use user defined variables.
Cartoony – cartoonybot@appspot.com – Replaces the text of every submitted blip with a cartoon balloon that contains the text instead. Colors the balloons based on username.
Dice Bot – dice-bot@appspot.com – Dice-rolling bot. Dice Bot will replace XdY (X is the number of dice; Y is the number of sides) with the results of those rolls.
Flippy – flippy-wave@appspot.com – Turns text upside-down.
Fnordlinky – fnordlinks@appspot.com – Replaces "PMID <number>" with article information from PubMed.
Hearty Emobot – hearty-emobot@appspot.com – Replaces ASCII art with wingding characters.
i-cron – i-cron@appspot.com – Evaluates Python expressions. Looks at blips in event, searches for CALC() macros and executes Python code using exec().
Insulty – megabytemb123@appspot.com – Information Needed
IPA Bot – ipa-bot@appspot.com – Changes normal letters into special characters used for phonetics.
Piratify – piratify@appspot.com – Turns whatever you type into "Pirate Speak" .. Arrrr.
Plotzie – plotzie@appspot.com – Plots sparklines from your data.
Shortee – Wish – Change “c u l8r” to “see you later” etc.
Swedish Chef – borkforceone@appspot.com – Changes english into Swedish-Chef Speak. Bork! Bork!
Syntaxy – kasyntaxy@appspot.com – Syntaxy does blip-by-blip syntax highlighting for a variety of languages including Python, Java, C, C++, html, css and javascript.
Watexy – watexy@appspot.com – Use LaTeX mathematical language in your Waves!
Wikify – wikifier@appspot.com – Replaces specific marked up text with a link to Wikipedia or a description relevant to the marked text.Games
Hangman – wavehangman@appspot.com – Play Hangman.
Roshambo – roshambowave@appspot.com – Play Roshambo (Rock / Paper / Scissors).
Speedy – Wish – Track the words per minute of all participants, competitive typing!Groups
Groupy – groupy-robot@appspot.com – Robot to manage groups.Integration
drop.io – mikeswaverobot@appspot.com – Creates a drop and puts the info into the wave whenever the robot is added as a participant.
OpenAustralia – In Progress – A robot to allow interaction with the OpenAustralia web site.
PlonieBot – In Progress – ploniebot@appspot.com – Brings wave document editing capabilities to the Plone CMS
Poppy – In Progress – poppywave@appspot.com – Helps bridge Google Wave conversations to email users outside the Wave.
Rssybot – rssybot@appspot.com – Turn google wave into an RSS reader!
Starify – starifybot@appspot.com – Lets you star waves, in sort of bookmarking style.
Tweety the Twitbot – tweety-wave@appspot.com – You can access your Twitter account.
Twiliobot – twiliobot@appspot.com – Transforms phone numbers into click-to-call links. If user clicks a link, a call is placed to his phone and to the number in the link. The call can be transcribed and inserted into the wave as text with a link to the audio.
Wave-Email – In Progress – wave-email@appspot.com – Provide an extension to Google Wave which will allow the integration of both sending and receiving emails.
Wave Live Messenger – wavelivemessenger@appspot.com – Allows you to chat to your windows live messenger contacts from inside a wave.Language
PhilBot – Wish – A suggested solution to the problem of waves with languages you can’t read.
Rosy Etta – rosy@wavesandbox.com – Translator (40 Languages).Polling
Polly the Pollster – polly-wave@appspot.com – Poll Bot.Search / Aggregation
Dr Maps – dr-maps@appspot.com – Updates a wave by inserting a map associated to an address.
Dr Weather – shiny-sky@appspot.com – Gives the weather for a City
Embedded Search Results – wave-sandbox@appspot.com – Web and Image searches inline.
FML Blipper – fmlblipper@appspot.com – displays random FML story from www.fmylife.com
Grauniady – grauniady@appspot.com – Searches the latest items from The Guardian for a given phrase.
Stocky – stocky-wave@appspot.com – Detects stock symbols from a wave and updates it with the live stock price.
Wavethingy – wavethingy@appspot.com – Searches Amazon for DVDs and books, and gives the author a cut of any purchases made off the links.
Yelpy – yelpful@appspot.com – Searches Yelp with a user defined location and category.Utilities
AmazonBot – amazon-withwaves-com@appspot.com – Enables social product research and shopping on Amazon.com. Wave participants can share products & reviews with contacts in real-time thanks to automatic queries by the AmazonBot against conversation keywords. The AmazonBot gadget can detect products and return inline product links or a custom full product browser.
Bloggy – blog-wave@appspot.com – Information Needed
Bit.ly Bot – bitly-bot@appspot.com – Shortens the url using bitly.
Botty – Wish – Will automatically add a set of useful bots to a wave according to a collection of bots (so they don’t have to individually be added when you use them all the time.
CountColon – countcolon@appspot.com – Adds text statistics to your blips (words, lines, etc.)
Companion Sphere – companionsphere@appspot.com – Collection of geek utils, first working verb is "lookup" for wikipedia/wiktionary one-line descriptions.
Databot – Wish – Will start as soon as the GData interface is published.
Emoticony – emoticonbot@appspot.com – Replaces text representations of emoticons with the relevant image.
JBREAKOUT – jbreakout@appspot.com – Debug utility that reports event triggers.
Maison – maison@appspot.com – Makes blips public at http://maison.appspot.com.
Multi – multi-wave@appspot.com – A quote collector. Reply a blip you want to quote with ‘quote this’ and randomly display a quote with ‘quote <wave @account.com>’. The bot is still being under development but you can try playing with it.
Natural Language Processing – knowledge-books@appspot.com – Adds blips with NLP analysis.
Nokar – lab2market@appspot.com – Has many features such as translations, image insertion, insert last tweets etc.
Posterous – posterous-robot@appspot.com – A robot for posterous.com user to post blog in Google Wave. Here is how to write a blog using Google Wave Robot for Posterous.
Publisher – wave-publisher@appspot.com – Information Needed
Skimmy – wave-skimmy@appspot.com – Converts text emoticons, from : ) to img. Has a bookmarklet which creates a popup menu to insert emoticons for which the code is unknown.
Smiley – smiley-bot@appspot.com – Changes the smiley symbols to smiley images.
Smiley – In Progress – smiley-robot@appspot.com – Changes the smiley symbols to smiley images.
Style Chart – stylechart@appspot.com – Inserts a chart into a wave.Wave Management
Bouncy – bouncy-wave@appspot.com – Allows you to remove robots from a wave. Doesn’t seem to work on real people though, and laughs if you try to ask it to kick itself out. To get it to kick a bot out, type "bounce:name@domain.com"
Linear – Wish – Enforce all replies to be to the main wave. If a user replies to a reply, remove it and place it as a reply to the main wavelet.
Read Onlie – readonliebot@appspot.com – Records the original wave content. Whenever it’s edited, the content is replaced with the original. Simple as that.
Seekdroid – seekdroid@appspot.com – You can list Robots, add them and find them out, easy to use. In continuous development. Website with all the information seekdroid.appspot.com.
Sweepy – sweepy-wave@appspot.com – Remove empty, whitespace-only blips.
Taggy – taggy-wave@appspot.com – Recognize #hashtags and add them as tags to the wave.
Tocgen – tocgen@appspot.com – Table of Contents auto-generated and updated based on the h1,h2,h3,h4 in a wave.
Twitusernames – twitusernames@appspot.com – Replaces all Twitter @username with links to the Twitter accounts.Gadget Utilities
Ajax Animator – In Progress – http://antimatter15.com/ajaxanimator/wave/manifest.xml – A fully integrated multi-user web based vector graphic animation authoring environment.
AmazonBot Gadgett – http://amazon-withwaves-com.appspot.com/gadgets/AmazonProductList.xml – The AmazonBot gadget can detect products and return inline product links or a custom full product browser.
Bidder – http://wave-api.appspot.com/public/gadgets/bidder.xml – Simple Auction.
Checky – http://wave-gadgets.appspot.com/checky.xml – Basecamp-like checklists with drag-and-drop.
Click me – http://wave-api.appspot.com/public/gadgets/hellowave.xml – Shows a button with a counter. Each time the button gets clicked, the counter is incremented by one. Shows off how the state interaction works.
HTML – http://wave-ide.appspot.com/html.xml – Embed any HTML into a wave.
iFrame – http://wave-ide.appspot.com/iframe.xml – Embed any web page into a wave.
iWave – http://gadget.wave.to/iWave/iWave.xml – Allows you to create a profile on wave to make wave just a little more personal. Uses facebook connect to retrieve your details if you sign in.
Licensing – In Progress – http://wave-license.appspot.com/license_gadget.xml – Creative Commons RDF Embedding – Planning Stage.
Maps – http://hosting.gmodules.com/ig/gadgets/file/101415471413908368316/mappy.xml -Embed Google Map.
Napkin – http://my-wave-gadgets.appspot.com/wave/NapkinGadget.xml – Example of Flash/Flex Wave Gadget, similar to Whiteboard gadget above – source on Google Code.
QuakeBot – In Progress – Server information on the Quake 3 protocol.
Raffly – http://raffly.googlecode.com/svn/trunk/sandbox/raffly-xml1/raffly.xml – Insert this gadget to select a random participant from your wave to be the winner. The winner of what? Well that’s up to you 🙂
Ratings – http://google-wave-resources.googlecode.com/svn/trunk/samples/extensions/gadgets/ratings/ratings.xml – Lets participants rate and review a topic (movie, restaurant, etc) in a wave and shows a tally of the result.
Slashdot Gadget – http://www.m1cr0sux0r.com/slashdot.xml – Loads latest Headlines from Slashdot.
Troco – An experimental peer-to-peer currency – http://troco.ourproject.org/gadget/org.ourproject.troco.client.TrocoWaveGadget.gadget.xml – Aims to provide a decentralized complementary community currency system, that is, a peer-to-peer currency system. Also you can see it as an IOU or promissory note based system. More info click here.
Vector Editor – http://jsvectoreditor.googlecode.com/svn/trunk/wave/vectoreditor.xml – A cross platform collaborative real time vector graphics editor.
Whiteboard – http://vps.michaelrose.id.au/canvas.xml – Draw on a virtual whiteboard.
Who is Coming? – http://wave-api.appspot.com/public/gadgets/areyouin/gadget.xml -Show a list of all people that have said whether they will come or not.Gadget Games
Backgammon – Wish – Remove all of one’s own checkers from the board before one’s opponent can do the same. [Wikipedia]
Battleship – Wish – Displays different board based on user.
Boxes – In Progress – Connect lines to make boxes and win.
Connect 4/Four-in-a-row – In Progress – sdunster@wavesandbox.com – http://www.sdunster.com/wave/four.xml – 2 users + observers, turn locking, just waiting to write win-detection code.
Floodit – http://gadget.wave.to/floodit/game.xml – 2 player race to fill a board with colors.
Magnetic Poetry – http://hosting.gmodules.com/ig/gadgets/file/107558585548952247431/fridge-11.xml – Re-arrange random words to form poetry.
Match them colors! – In Progress – Match 3 / gem matching game.
Othello – Wish – Play Reversi.
Sudoku – http://blah.appspot.com/wave/sudoku/sudoku.xml – Play Sudoku.
The Button – http://hyperthese.net/wave-gadgets/the-button.xml – A useless (I mean USELESS) game.Hooks
CVS integration – Wish – CVS history can be converted into a wave with playback.
GIT integration – Wish – GIT history can be imported and played back (dffs).
SVN integration – Wish – SVN History can be converted into a wave with playback.Appearance
Google Wave Scrollbars – http://www.uniformedopinion.com/google-wave-native-scrollbars-extension/google-wave.crx – Changes the wave scrollbars to the default system scrollbars.
Enhanced Google Maps. In case you haven’t noticed, we’ve added yet another Google Map to Nerd Vittles. Now, in addition to showing our location with Google Latitude, we also are displaying your location based upon your IP address. We’ll show you how to add something similar to any LAMP-based Linux system in coming weeks. It’s a powerful technology that has enormous potential. If you’re unfamiliar with Google Maps, click on the Hybrid and Satellite buttons and then check out the scaling and navigation options. Double-click to zoom. Incredible!
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
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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…
For those of you that already have Google Wave accounts, here’s a sample of how a wave will look in a WordPress posting. You can even add content to the wave! This works in Safari and Chrome most of the time, Firefox some of the time (if you turn on Accept 3d Party Cookies), and IE almost never. For those of you that are not part of the Google Wave preview, you’ll just have to wait patiently until Google turns on at least read-only access to this functionality:
Asterisk Virtualization: PiaF + Proxmox, It Just Works
We've invested weeks and months over the years wrestling with virtualization technologies searching for the perfect fit for the Asterisk® PBX platform and especially for the turnkey solutions provided by PBX in a Flash and our latest Orgasmatron V installer. Why virtualization you might be asking? As with most computer applications, it comes down to flexibility and, of course, cost savings.
For the latest article on PBX in a Flash 2 with OpenVZ, follow this link.
In the flexibility department, VoIP virtualization lets you choose options such as Cloud Computing and hosted solutions from various providers. It also provides a terrific training platform as well as your own managed Cloud Computing solution. You can build and host a dozen or more virtual Asterisk systems on a single $500 to $1,000 server and have a transportable solution ready to deploy in a couple of hours. And then there are those of us in the technology business that need to test all sorts of new operating systems and applications without having to dedicate a standalone machine to each experiment.
Security WARNING: Always run Proxmox behind a secure, hardware-based firewall with no port exposure to the Internet. Review this message thread for the reasons why.
Our virtualization platform of choice is Proxmox, a lightweight Debian-based distribution that includes kernel support for both KVM and OpenVZ. As Martin Maurer from Proxmox put it in a recent interview:
This means you get the best of both virtualization worlds... containers (OS Virtualization) and fully-virtualized machines (Machine Virtualization). Proxmox VE also includes a very powerful yet easy to use web-based management system with clustering features. Boot the Proxmox VE install media, answer a few simple questions, and within 10 minutes you have a very powerful virtualization platform you can manage from a web browser. Install it on one or more additional machines that are networked together and use Proxmox VE's cluster management tool to create a virtualization cluster that allows for centralized management, automated backups, iso media and OS Template syncing, as well as migration features. Proxmox VE really is a time saving turnkey solution... and it is freely available under a GPL license."
As far as cost savings, $500 to $1,000 says it all. When you can run a dozen dedicated systems on such a hardware platform, it reduces the individual cost of each turnkey system deployment to well under $100. And the performance penalty for implementing this multitasking solution is only a 1 to 3 per cent performance hit compared to using comparable standalone systems for similar computing tasks. Om Malik recently noted that:
More than half of new servers in 2009 will be virtualized, compared with 30 percent in 2008, according to a new survey by TheInfoPro."
Comparing 2009 to 2008 deployments, that's a 70% increase in just one year. When there is comparable performance, 90% cost savings, and greatly enhanced deployment flexibility, you have to ask yourself why wouldn't you deploy virtualized solutions. With the solution we're providing today, you get some other benefits as well: snapshot backups and cluster computing, both of which actually work. And the cost of this virtualization technology... it's FREE!
Hardware Requirements. For full KVM virtualization support, you'll need either an Intel-VT1 or AMD-V2 capable CPU/Mainboard. Also strongly recommended are a multi-core CPU and as much RAM as your budget can afford. Our favorites (primarily because of cost) are the Dell T105 (with either dual or quad core AMD Athlon processor) or the Dell T300 (with quad core Intel Xeon processor). Both are on sale for the next few days starting at $249 up to about $1,000 with $350-$549 off the retail prices. You can save more by using our Dell coupon in the right margin. We recommend purchasing larger hard disks from other suppliers so stick with the default setup in drives. Dell has gotten more competitive on RAM pricing so that's your call. For a point of reference, a dual core AMD with 8GB of RAM can support about 8 simultaneous Asterisk servers.
Installing Proxmox. If you go the Dell route, you'll need an external USB CD or DVD drive to install Proxmox. Dell's optical drives aren't supported in the Proxmox boot image. So begin by downloading the Proxmox VE 1.3 ISO image and create your CD. Then boot your new server from the CD (by pressing F11 for the boot selection screen and choosing your USB external drive on Dell servers). Press Return to begin the install, agree to the license agreement, and click Next on the installer screen to begin. Choose your country, time zone, and keyboard layout. Next choose a secure password and provide a valid email address which is used to send you critical alerts from your Proxmox server. Finally, choose a hostname, specify a fixed IP address, netmask, gateway, and DNS servers and then press Next. Three minutes later, you'll have a new Proxmox server. Log in to your server as root and create a directory for your backups: mkdir /backup. You're finished on the CLI at this point.
OpenVZ vs. ISO Images. One of the beauties of Proxmox is that it supports two different types of images to create virtual machines. An OpenVZ template is akin to a snapshot of an existing system while an ISO image is identical to the installer you normally would burn onto a CD in order to install a software application on your server. In short, you still have to go through the installation scenario when you create a virtual machine (KVM) from an ISO image. A virtual machine created from an OpenVZ image is ready for use the moment it is created. If you remember when instant-on televisions first were introduced, you'll also appreciate the difference in boot times between OpenVZ and KVM machines which boot an application installed from an ISO in much the same manner as you would experience on a standalone machine.
As with life, there's a dark cloud lurking behind every silver lining, and this is especially true in the Asterisk environment. OpenVZ containers rely upon a shared kernel, the one that actually boots the Proxmox server. KVM containers created from ISO images are self-contained with their own complete operating system and kernel. Thus, zaptel and dahdi cannot be loaded directly from an OpenVZ container. Instead one must rely upon a shared version of zaptel or dahdi loaded on the Proxmox server itself. As it turns out, this is no small feat and certainly not a task for mere mortals. Bottom Line: If you need conferencing or otherwise need a timing source for your Asterisk deployment, you will not want to use the OpenVZ approach at least for now. We hope to more fully document the zaptel/dahdi hurdles that need to be addressed in coming weeks. You can follow our progress in this message thread on the PBX in a Flash Forum. On the other hand, if you have more traditional VoIP requirements for your PBX, then the ease of installation and use of the OpenVZ image makes perfect sense. So let's start there assuming you understand the limitations.
Installing PIAF OpenVZ. Using a web browser, download the new PBX in a Flash OpenVZ image to your Desktop. Our special thanks to Wolf Paul, who did most of the work in putting this together. Once you have the OpenVZ image in hand, point your web browser to your Proxmox server: https://ipaddress. Accept the default certificate and login as root. You'll get a Welcome screen that looks something like what's shown above. Click on the Appliance Template option. In the Upload File section, choose the PIAF OpenVZ image on your Desktop and click Upload. Be patient. It's a big file. So go have a cup of coffee. You'll get a prompt when it's completed. And, as Joe Roper has pointed out, you can do this directly within the Proxmox server by logging in as root and issuing the following commands. Thanks, Joe.
cd /var/lib/vz/template/cache/
wget http://tr.im/piaf1506
If you really want to walk on the wild side, here's a third method from Ap.Mathu. After logging into your server as root and issuing the following commands, you can download PBX in a Flash as well as Joomla!, eyeOS, BlueOnyx, Moodle, and FrontAccounting directly through the Proxmox web interface (Appliance Templates, Download):
cd ~
wget http://mundy.org/piaf1506
cat piaf1506 >> /var/lib/pve-manager/apl-available
NOTE: You'll need to use the third option above only after you enable IPtables below because the apl-available file gets regenerated from "headquarters" each time Proxmox restarts.
Enabling IPtables Firewall. IPtables works a little differently in the OpenVZ environment. It actually runs on the Proxmox host. There are three steps to get it working. First, be sure you have downloaded PIAF OpenVZ template 15.04 or later. Second, shut down every running VM on your Proxmox server using the web interface. When you're sure they're all stopped, log into your Proxmox server as root using SSH and carefully enter the following two commands. Note that, because of the length, the sed command stretches to several lines which should be unraveled into a single line for the command to execute properly! Using a block-copy from a desktop machine to your SSH session is the safest method.
sed -i 's|ipt_REJECT ipt_tos ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length|ipt_REJECT ipt_tos ipt_TOS ipt_LOG ip_conntrack ipt_limit ipt_multiport iptable_filter iptable_mangle ipt_TCPMSS ipt_tcpmss ipt_ttl ipt_length ipt_state iptable_nat ip_nat_ftp|' /etc/vz/vz.conf
/etc/init.d/vz restart
Now you're ready to create your first virtual machine. Click on Virtual Machines and then the Create tab. Accept the default OpenVZ container type and give your virtual machine a host name that will help you distinguish it from other VMs on your Proxmox server. Create a secure root password for your new VM. We recommend a minimum memory and swap memory size of 512MB and a minimum disk size of 20GB. You can experiment with these to find the best fit on your server. It only takes about 30 seconds to create an OpenVZ virtual machine so trial-and-error isn't painful.
You have a choice of Network Types. With Virtual Networks (venet), you need to designate a static IP for your virtual machine. With Bridged Ethernet (veth), an IP address is assigned by your DHCP server. Be aware that our status app currently won't display venet-assigned IP addresses, but ifconfig will. There are some other significant differences including network security that you may wish to review. Our special thanks to Martin Maurer from the Proxmox Dev Team for the hand-holding in getting both options working. To keep things simple, choose Bridged Ethernet as shown in the screen shot above. As mentioned, we'll depend upon your DHCP server to assign a dynamic IP address. You can lock it down on your router to assure that the same IP address always is assigned to this virtual machine. Finally, provide a DNS domain for the new VM and assign at least one DNS server. The IP of your gateway router/firewall usually will suffice. Click create when you have filled in all the blanks. Your new virtual machine will be ready to run in less than a minute.
To start the OpenVZ virtual machine, click on the List tab. Then click on the VM you wish to run. When the details display, click the Start button. Within a couple seconds, your VM will start up. Now click on the Open VNC Console link which provides you a command line interface to the now running virtual machine. Type ifconfig several times until you get a display showing your network interfaces. If no IP address is shown for eth0, type: service network restart. You only need to do this the first time your new virtual machine is started. Once the network reloads, you should be good to go. Type status and the IP address of your new VM should display. Type service iptables status to verify that IPtables is running. It currently does not show properly with status. If it's not running, type service iptables restart, and then check it again. The safest test is to attempt to log into your new server with a phone using the wrong extension password. After three tries, it should lock out that IP address temporarily.
Now it's time to secure your new virtual machine. We need to change the master password (not the root password) that is used to gain web access to your server. We also need to change the server's SSH keys to make them unique. Just run the following three commands making certain that you choose to overwrite your existing SSH keys when prompted to do so:
passwd-master
ssh-keygen -f /etc/ssh/ssh_host_rsa_key -t rsa
ssh-keygen -f /etc/ssh/ssh_host_dsa_key -t dsa
Finally, you can type rasterisk to load the Asterisk CLI. You now have a functional PBX which is ready for configuration. See our knol for step-by-step instructions if you're new to all of this. Or, better yet, you can transform your new virtual machine into a turnkey PBX in less than 10 minutes with free calling in the U.S. and Canada with our Orgasmatron V Installer.
We strongly encourage (actually we're begging) you to read our Primer on Asterisk Security before doing anything else. It could save you an astronomical phone bill down the road.
Where To Go From Here. Until our next chapter, you might want to experiment with some of the other OpenVZ appliances which are available for Proxmox. Many can be installed within the Proxmox GUI (Appliance Templates, Download). Here's the short list: Proxmox Mail Gateway, CYAN Secure Web, Trouble Ticket Tracking, Zenoss Core IT Monitoring, CentOS 4 and 5, Debian 4 and 5, Fedora 9, Ubuntu Hardy, Drupal Content Management, Joomla Content Management, MediaWiki, SugarCRM, and WordPress. Enjoy!
Continue reading Part II for the 64-bit version with DAHDI conferencing...
Enhanced Google Maps. In case you haven't noticed, we've added yet another Google Map to Nerd Vittles. Now, in addition to showing our location with Google Latitude, we also are displaying your location based upon your IP address. We'll show you how to add something similar to any LAMP-based Linux system in coming weeks. It's a powerful technology that has enormous potential. If you're unfamiliar with Google Maps, click on the Hybrid and Satellite buttons and then check out the scaling and navigation options. Double-click to zoom. Incredible!
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
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...
- Be very careful choosing Intel processors. Even some high-end processors do not support Intel Virtualization Technology. Here's the official list. [↩]
- And here is a useful reference for AMD-compatible processors. The AMD WIKI provides the following list of AMD-V compatible processors: "AMD's x86 virtualization extension to the 64-bit x86 architecture is named AMD Virtualization, also known by the abbreviation AMD-V, and is sometimes referred to by the code name 'Pacifica'. AMD processors using Socket AM2, Socket S1, and Socket F include AMD Virtualization support. AMD Virtualization is also supported by release two (8200, 2200 and 1200 series) of the Opteron processors. The third generation (8300 and 2300 series of Opteron processors) will see an update in virtualization technology..." [↩]
VoIP Prioritizing The World’s Best Traveling Phone
We follow a lot of really smart geeks on Twitter. As you might imagine, there’s a good bit of chatter about the world’s best cellphones. About half are die-hard iPhone users, and the rest are all over the map. Our iPhone is now a glorified iPod and, when you finish reading today, you’ll understand why.
What always has set Macs apart from PCs in our humble opinion is flexibility. So why is it that Apple has gone out of its way to strip that feature from the iPhone? Well, we all know the answer. AT&T and the iTunes Store. Or in a word, money. So what’s missing? For openers, there’s no tethering, the ability to connect your PC to your cellphone when the power goes out so you can send an emergency message or check on your servers at work. And then there’s free calling: the ability to place free SIP calls or Google Voice calls using your cellphone from almost anywhere. And then there’s the money thing. If you’ve traveled to foreign countries with an AT&T-powered iPhone, we don’t have to finish this story. For everyone else, let’s just say the cost of using your iPhone in a foreign country or on a cruise ship is stratospheric.
We’ve watched our friends and colleagues purchase all sorts of add-on gizmos to make up for the shortcomings in the iPhone. These have included secondary cellphones and more recently the MiFi devices which let you pay one of the companies in the American cellphone oligopoly another $60++ per month to tether your notebook and netbook to the cellular data network. Let’s get this straight. We pay a cellphone provider for an unlimited data plan as part of our service, but to transmit data to or from our PC through the plan, add another $60 a month for another data plan with a bandwidth cap. Huh? This is for a service that most of us use intermittently and would prefer to never use because of the lousy performance. Here’s our #1 traveling rule. Never stay in a hotel that doesn’t have WiFi, period. Why would you? The one next door has it!
So let’s go about this by the book… with a requirements analysis first! We want a cellphone that makes cellular calls from most locations, and we want the ability to decide which cell provider we use depending upon where we are. We want the option to make phone calls through our own SIP provider, or Asterisk® server, or Google Voice whenever we feel like it with or without a Wi-Fi connection. And, of course, we want VoIP Prioritization. This means we want our cell phone to prioritize incoming and outgoing calls by attempting to use VoIP services first, cellphone carrier second. Good luck with that one! We also want to be able to check our email using POP3 or IMAP servers. And, when we need to send or receive something on our notebook computer and there’s no WiFi around, we want our cellphone to provide data connectivity. We’re not going to be downloading movies and 1,000-page books all day long. We just want to get an important file attachment from the office so we can read it on a normal screen. And, finally, we’d like a QWERTY keyboard for messaging, and we want to be able to change our own battery, add a memory chip, and swap out SIM cards whenever we’d like. And the music, camera, and GPS functionality would be nice-to-haves on a phone.
Is this so hard? Well, if you’re in the United States and you’re planning to purchase a phone through Sprint, T-Mobile, AT&T, or Verizon to get one of those sign-away-your-life phone discounts, the answer is IMPOSSIBLE! And, to those that are chomping at the bit to tell us how they’ve accomplished some of these miracles with their hacked iPhone, let me just remind you that Apple considers it a national security threat to hack your iPhone thus explaining why Apple also considers it honorable to brick your hacked iPhone at any time despite the fact that you paid for it. Ask yourself if you really want to invest your cellphone dollars with a company spewing forth this kind of bullshit stuff.
And the answer is…
The unlocked U.S. version of the Nokia E71 costs $289.99 at NewEgg, and it’s worth every penny. We’ve been using ours all day, every day for the better part of a year. We’re not going to do a full review of the phone when there’s already an excellent one out there. Start with the allaboutsymbian review and then pick up again here. What isn’t covered in that review is the critical component that we believe sets this phone apart from everything else out there: incredibly simple SIP connectivity and VoIP setup with an Asterisk server because of the native SIP stack and SIP client which is built into the E71’s firmware. And, as you will soon discover, this transforms the E71 into the perfect traveling companion because it makes the E71 just another telephone extension on your home office Asterisk PBX. If secure communications matters, there’s VPN support as well.
Implementing Incoming VoIP Prioritization. Here’s how we’ve set up connectivity to our E71. First, create an extension on your Asterisk server that will be dedicated to remote SIP access from your E71. Let’s use extension 371 in this example. Give it a very secure password because the IP address of your E71 will change as you move from place to place so we can’t really lock down the extension with anything other than a secure password, or you won’t be able to connect. Next, create another extension (372) and forward all incoming calls to that extension to the regular phone number of your E71, i.e. the one provided by your cellphone provider. Then create a Ring Group on your Asterisk server (373) and set up 371 as the only number in the ring group extension list. For the destination if no answer, choose extension 372. Finally, set up your Google Voice number with a destination extension that forwards calls to ring group 373. So the way this will work is that incoming calls to your Google Voice number will ring the SIP connection on your E71 (371) if your E71 is registered to your Asterisk server via SIP. And, when it’s not registered, the calls will be forwarded to the regular phone number of your E71 (372) without any delay since extension 371 isn’t registered with your server. If you get in the habit of searching for WiFi wherever you happen to light and connecting back to your Asterisk server, (as you’ll see, this is a one-click operation), then you’ll have dirt-cheap remote cellphone service on your E71 almost all of the time. And, if you travel to foreign countries, it means that any time your E71 is registered with a WiFi HotSpot, all incoming calls will be free instead of costing an arm-and-a-leg in per minute international roaming fees.
SIP Setup for Nokia E71. John Rogers over at geek.com has written an excellent piece with lots of pretty pictures to show you how to configure your E71 with Asterisk. Rather than reinvent the wheel, here’s the link. It only takes a couple of minutes. We do have a few tips to get you started on the right foot. Make certain that the IP address you enter for your Asterisk server is the public IP address or fully-qualified domain name for your server, not the private IP address inside your firewall. As you roam from one WiFi network to the next, the E71 will automatically configure the phone for the new networks as soon as you choose WLAN Scanning, select a WiFi network, and choose to Connect to your Asterisk server. This is performed from the default screen on your phone so there’s no wading through layer upon layer of menus. After linking and unlinking to different networks about a dozen times, we have found it’s a good idea to shut down the phone, remove the battery momentarily, and then restart the phone. It keeps awkward connect problems from ever occurring. To enable VoIP Prioritization for outbound calling, all you have to do is change one default setting on the Nokia E71: Menu, Tools, Settings, Phone, Call, Default Call Type: Internet Call.
Depending upon your choice of router, using the public IP address of your Asterisk server may cause connectivity issues when you attempt to make a connection through the same WiFi network on which your Asterisk server resides. You can solve this by investing in one of dLink’s Gaming Routers which also provide the necessary tools to prioritize VoIP traffic on your network. Second, make sure you load the latest Nokia firmware for the E71 before you begin configuring your phone. You can check which firmware is installed on your phone by pressing *#0000#. If it’s less than 200.21.118, you need to upgrade, and you’ll need a Windows machine to do it. Here’s the link to Nokia’s upgrade site.
Where To Go From Here. Once you have your E71 performing as a remote Asterisk extension, there are some other must-have’s for your phone. First, you’ll want to purchase JoikuSpot Premium for 15.00€ (about $20). It turns your phone into a WiFi HotSpot whenever you need tethering. Next you’ll want to load Nokia’s OVI store which includes a number of free downloads including Internet Radio, Fring, Nimbuzz, and Web Server. With the web server, you can actually create a blog and let visitors share photos and take pictures using your E71. Try ours to get a taste of what’s available. We think you’ll also find Google Latitude to be a fascinating addition. It lets you produce a free, GPS-enabled map with your current location just like Where In the World Is Nerd Uno. In fact, that map is produced from GPS data generated on our Nokia E71.
A Word of Caution. Finally, we’ll close on a cautionary note. Tempting as it may be to buy Nokia’s latest and greatest cellphone, DON’T! Nokia quietly has dropped the native SIP stack and SIP client on almost all of its newest cellphones presumably to win the love and affection of companies like AT&T. These are the same companies that continue to claim in FCC filings that they have nothing against VoIP on cellphones. The list of VoIP-impaired Nokia cellphones includes the N97 as well as the AT&T-branded E71x. Nokia also has been less than clear about the new N900. Historically, this has meant that SIP functionality has disappeared. So beware of shiny new things… that may not work worth a damn. It’s too bad. Nokia was one of our favorite companies, but it looks like they’re ceding the VoIP technology business to Google’s Android which happens to be next on the Nerd Vittles Radar. Here’s a complete list of Nokia’s SIP-compatible phones. Enjoy!
Enhanced Google Maps. In case you haven’t noticed, we’ve added yet another Google Map to Nerd Vittles. Now, in addition to showing our location with Google Latitude, we also are displaying your location based upon your IP address. We’ll show you how to add something similar to any LAMP-based Linux system in coming weeks. It’s a powerful technology that has enormous potential. If you’re unfamiliar with Google Maps, click on the Hybrid and Satellite buttons and then check out the scaling and navigation options. Double-click to zoom. Incredible!
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
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…
Tweaking Asterisk for Free Google Voice Calling
Now that the Asterisk® and Google Voice marriage is finally underway, we wanted to step back today and revise the original methodology a bit to take advantage of some of the terrific comments which were offered in response to our last article. First, the good news. U.S. calls through Google Voice using Asterisk work! They sound great, and they're free. The not so good news was that the MeetMe conferencing trick to join your outbound call with the Google Voice click-to-dial return call from your destination worked great so long as a real person answered the phone. But, if an answering machine picked up or no one answered the call at all, there were problems because these calls already had been transferred to the MeetMe conference and there was no simple way to disconnect them. And the need for two DIDs to support a single Google Voice interface just seemed a bit wasteful.
9/1/2010 Update: A good bit has changed with Google Voice since this article was first published. For the definitive guide and installation procedure, we highly recommend The Incredible PBX and accompanying article which can be found at this link. Google Voice (and much more) already is included in our new PBX which is literally Plug-and-Play. If you prefer to roll your own, be sure to also have a look at this excellent update on the Michigan Telephone Blog.
Today we want to try to eliminate these two quirks while stiill providing a seamless interface between Google Voice and Asterisk. We also appreciate that thousands of you already have implemented the previous approach. So we want your transition to the new way of doing things to be as painless as possible. On the other hand, for frequent readers, we hope you'll bear with us as we repeat some of what already has been covered in previous articles so new visitors don't have to jump around between articles to get the complete picture of what we're trying to accomplish.
The objective remains the same. We want a methodology that lets us make outbound calls from any Asterisk phone using the Google Voice service to take advantage of free calling in the United States and Canada. And we want calls to our Google Voice number delivered to our Asterisk system for transparent call processing. Yes, SIP is still on our wish list for both outbound and inbound calls with Google Voice, but we'll make do with PSTN calls particularly while Google is footing the bill for all of the calls.
Update: There's now a turnkey Asterisk solution that implements Google Voice calling without getting your hands dirty. Check out our new Orgasmatron V.
Tweaked Design. Here's the new design. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking a few days from application to invite which is really great news. Don't use a space in your Google Voice password! Once you have a Google Voice account and phone number (Google has reserved a million of them so... not to worry!), then you'll need a DID that provides unlimited, free incoming calls. Once you get your DID set up on your Asterisk system, we'll set up a forwarding phone number for this DID in your Google Voice account so that Google Voice calls can be connected to your Asterisk server.
For outbound calls, we'll combine a little dialplan voodoo with pygooglevoice to instruct Asterisk to place a click-to-dial call using your Google Voice forwarding number. Then we'll stuff in the destination U.S. phone number. When you dial GV-678-1234567 from any of your Asterisk phones, Asterisk will park your initial call in a reserved parking lot slot and then join the called party to the originally parked call. The entire procedure is virtually transparent both to the caller and the callee. And, unlike the MeetMe conference, the parking lot fades out of the picture as soon as the call is connected. Thus, if either party hangs up, the active channel for the call is terminated on your Asterisk server.
For inbound calls from your Google Voice number, we'll tweak the dialplan so that it can distinguish between a RingBack call that Google Voice initiated and a true inbound call. We'll peel off the real inbound calls and route them to a separate Inbound Route in FreePBX for processing in any way you desire.
Finally, for those that implemented the methodology in our previous article, we'll walk you through the steps to revise your existing setup to take advantage of these new tweaks. You can skip over the initial installation process if you already have gone through the Google Voice setup from our earlier article. Just skip down to Tweaking Previous Setups.
Special Thanks. At the outset, we again want to express our sincere appreciation to Jacob Feisley and Paul Marks for their pioneering work on a Python interface to Google Voice. We also stumbled upon another Python development project, Google Voice for Python. While we originally had planned to rely upon Jacob and Paul's script, we ultimately decided to implement pygooglevoice because of the additional flexibility it provided for down the road. With pygooglevoice, you not only can make Google Voice calls, but you also can send SMS messages with no muss or fuss. Jacob Feisley has now joined that project as well. So, our special tip of the hat goes to the entire Google Voice for Python development team. It's a terrific product as you will see.
Prerequisites. Today's setup requires a CentOS-based Asterisk aggregation with a current version of FreePBX. Be aware that today's solution requires Python 2.4 or higher and reportedly will not work with Python 2.3 found in some Linux distributions. We've tested everything with PBX in a Flash and, on that platform, you're good to go. The install script should work equally well with the other CentOS-based Asterisk aggregations, but we haven't tested them. Be our guest, and let us know if you encounter any problems. Finally, a word of caution. We don't ordinarily distribute solutions using development tools we don't use. Our knowledge of Python wouldn't fill a thimble. We've made an exception today because of the extraordinary interest in Google Voice by the Asterisk community. But, if something comes unglued, we can't fix it. So have a backup plan in place just in case. 🙂
Today's Drill. To get everything working today, there are six steps: (1) obtaining and configuring a DID to manage calls between Google Voice and Asterisk, (2) configuring a Google Voice forwarding number for this DID to manage your outbound and inbound calls, (3) configuring FreePBX to route all outbound calls with a GV prefix to your special Google Voice dialplan context, (4) configuring an inbound route to manage incoming calls from your Google Voice number, (5) setting up a series of Parked Call extensions, one of which will be used to manage your outbound Google Voice calls, and (6) running our install script which adds the dialplan code for Google Voice calling with your credentials and puts the Python application into place on your server. It sounds more complicated than it is. So hang on to your hat. Here we go!
Dedicated DID. Before you can use Google Voice with Asterisk, you'll need a DID that can be dedicated to your Google Voice interface to Asterisk. We'd recommend a free IPkall or SIPgate DID. To get started, use one of the links above to obtain and configure the DID. Temporarily point the DID to an extension on your Asterisk system that can be used to verify your requests for the number. Since all of these calls are free, the area code of the DID really doesn't matter because you're never going to publish the fact that it exists.
The easiest method for setting up the DID is to first create a SIP URI for the DID on your Asterisk system. Next route the SIP URI to an Inbound Route in FreePBX where you can manage the destination for calls to that DID. Initially, you want the destination to be an extension on your Asterisk system that you can answer to verify both the DID setup and the GV setup below. Finally, point the DID you obtained to the SIP URI defined above.
HINT: The entry in extensions_override_freepbx.conf would look something like this for a SIP URI called ipkall-1:
exten => ipkall-1,1,Goto(from-trunk,${DID},1)
Then you would create an inbound route named ipkall-1 using FreePBX and designate some existing extension on your server as the destination for these inbound calls.
When you set up the SIP forwarding for the DID at ipkall.com, you'd specify the SIP URI as:
ipkall-1@ipaddress_of_your-Asterisk_server
We've previously covered in detail how to do this so read the article if you need a refresher course. To reiterate, the area code of this DID really doesn't matter because you're never going to give out the number. So use one of the free sources and save yourself some money. The real trick is you want to use a DID with unlimited, free inbound calls. Both IPkall and SIPgate provide that functionality at no cost.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of your DID. Be sure the DID is pointed to an extension on your PBX that you can answer since you have to go through Google's confirmation drill to successfully register the number. After the DID is confirmed, be sure there's a check mark beside this Google Voice destination so that incoming calls to your GV number will be routed to your Asterisk server.
While you're still in the Google Voice Setup, click on the General tab. Uncheck Enable Call Screening. Turn Call Presentation Off. And set CallerID to Display Caller's Number. Be aware that IPkall DIDs only forward your IPkall number as the CallerID number while SIPgate DIDs reportedly forward the actual number of the person calling you. If this matters to you, then you may prefer the SIPgate DID option. Finally, uncheck Do Not Disturb. Now click the Save Changes button.
Integrating Google Voice into Asterisk with FreePBX. Open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert your GV number in the Outbound CallerID field and add the following Custom Dial String on the form and Submit Changes and reload the dialplan:
local/$OUTNUM$@custom-gv
Next, choose Setup, Outbound Routes, Add Route and fill in the following entries on the form:
Route Name: GoogleVoice
Dial Pattern: 48|NXXNXXXXXX
Trunk Seq: local/$OUTNUM$@custom-gv
Inbound Routes. Next, we need two Inbound Routes to get everything working. In setting up your DID with IPkall or SIPgate, you already should have created one inbound route for that provider. It already should be routing calls to an extension on your PBX. Now we need to create a Custom Destination for this inbound route and then reroute these calls there. In that way, your RingBack calls will be routed to some special dialplan code that drops these calls into a custom parking lot where the RingBack call is married up to the extension from which you placed the original call. Then we need to create another inbound route to manage normal incoming calls that are forwarded to your PBX whenever someone dials your Google Voice number.
To begin, choose Tools, Custom Destinations, Add Custom Destination and add an entry like this and then click the Submit Changes button:
Custom Destination: custom-park,s,1
Description: Custom GV-Park
Next choose Setup, Inbound Route and click on the inbound route you created previously for IPkall or SIPgate. Change the destination for these calls to Custom Destination: Custom GV-Park.
Now click on Add Incoming Route and create a new route for your incoming Google Voice calls. Give it any description you like but, for the DID number, it must be gv-incoming. You can leave most of the other defaults. Just be sure you set a destination for your incoming calls from Google Voice. It could be an extension, ring group, IVR, or whatever best meets your needs. The important entry here is gv-incoming for the DID number. Click the Submit button to save your entries. Ignore the warning that you've entered an oddball DID. We know what we're doing. 🙂
Setting Up the Parking Lot. While still in FreePBX, we need to create or adjust your existing settings in Setup, Parking Lot. The parking lot is used by FreePBX to simulate old key telephones where you could place a call on hold and then someone else in the office could pick up the call by clicking on the blinking key on their phone. The Asterisk equivalent is to press the flash hook and dial your Parking Lot Extension which then places the call in a Parking Lot space and tells you what the space number is. Someone else then can dial the number of that space to pick up the call. Our little trick today works like this. When you place an outbound call through Google Voice, your extension will be dumped into a reserved parking lot space. When Google Voice initiates the RingBack call before connecting the destination number you've dialed, that call will be sent to the same reserved parking lot space. The two calls then are joined, and you'll hear the parking lot number followed by ring tones as your call is connected by GV to its final destination. Our special thanks to Richard Bateman for his comment on the previous article and this terrific tip! He wins an Atomic Flash installer from Nerd Vittles. In addition, A. Godong wins an Atomic Flash installer for his tip on consolidating two DIDs into a single DID to manage both inbound and outbound GV calls. Just send us your addresses.
Now, where were we? Most FreePBX systems have a default setup for the Parking Lot. What we need to do is be sure you have reserved one more space in the parking lot than you actually need for day to day operation of your PBX. We'll use the last parking lot space number to manage outbound calling through Google Voice. Our entries look like the following:
Enable Parking Lot Feature: checked
Parking Lot Extension: 70
Number of Slots: 5
Parking Timeout: 30 seconds
Parking Lot Context: parkedcallsDestination for Orphaned Calls: Terminate Call: Hangup
If you use our setup above, the Magic Number is 75 which is the fifth slot in the Parking Lot. If you use a different Parking Lot extension or number of slots, here's how to calculate the Magic Number. Start counting the slots beginning with one more than the Parking Lot Extension. When you get to the last slot in the number of slots you've specified, that's your Parking Lot Magic Number. Write it down. You'll need it in a second when you run our GV installation script.
Save your entries and reload the Asterisk dialplan when prompted.
Integrating pygooglevoice. Now we're ready to complete the setup by running our revised script which loads pygooglevoice and sets up your dialplan in extensions_custom.conf. You'll need 5 pieces of information to run the script so write them down before you begin:
1. Your 10-digit Google Voice phone number
2. Your Google Voice email address
3. Your Google Voice password (no spaces!)
4. Your 11-digit RingBack DID (16781234567)
5. Your Parking Lot Magic Number
A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).
Now log into your Asterisk server as root and issue the following commands:
cd /root
wget http://bestof.nerdvittles.com/applications/gv/install-gv-new
chmod +x install-gv-new
./install-gv-new
Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.
exten => 333,1,Dial(local/6781234567@custom-gv,300)
Congratulations! You now have what we hope will be flawless and free U.S. calling on your Asterisk system using Google Voice. No gimmicks, no strings, no cost. Enjoy!
Finally, one additional word of caution. Both Google Voice and this call design are set up for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service.
Asterisk 1.6 Solution. Several readers now have documented the procedure for implementing the Asterisk 1.6 bridge technology to make outbound Google Voice calls. You can read all about it here.
Tweaking Previous Setups. If you installed pygooglevoice using our previous tutorial, here's what you need to do. First, log into your Asterisk server as root and issue the following commands:
cd /etc/asterisk
nano -w extensions_custom.conf
Scroll to the bottom of the file by pressing Ctrl-W then Ctrl-V. Move up the file using up arrow until you reach [custom-gv]. Press Ctrl-K repeatedly to delete all of the lines in the [custom-gv] context. If you get to another line that starts with a label in brackets like [this], STOP deleting. Once you've deleted all of the lines in the [custom-gv] context, save the file: Ctrl-X, Y, and press Enter.
Now continue reading this article by jumping up to the Google Voice Setup topic. The Custom Trunk entry and the GoogleVoice outbound route will already be in your FreePBX system so there's no need to repeat those two steps. You will need to perform the remaining FreePBX steps beginning at the Inbound Routes topic and continuing on with Setting Up the Parking Lot. Finally, when you run the new installation script, it will detect that pygooglevoice is already on your system and will skip that step but will install the new custom contexts in extensions_custom.conf using your new settings. Enjoy!
Thought for the Day. Which is more arbitrary: (1) Apple snubs Google Voice or (2) Google Voice snubs SIP? Pays to look in the mirror occasionally.
Best Read of the Week. Memo to Steve Jobs and Apple: Stop Being A Jerk!
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
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 Wedding Is On: Asterisk and Google Voice Tie the Knot
Well, it was another painful week on the Google Voice front for Asterisk® lovers. We would hasten to add that none of the pain was induced by Google. This time around it was Gizmo5 initiating the daily gear shifting. What started out last Monday as a free, unlimited Google Voice service quickly morphed into a 20 minute call, and then a 3 minute call, and then 2¢ per minute for Google Voice calls. No notice to the early adopters, of course. So they'd only learn about all of this when the funds in their Gizmo5 accounts were exhausted. Ouch! We have gotten an explanation from Michael Robertson at Gizmo5 as have others. But we're not in a position to verify any of the information so we're not inclined to just pass it on again. Burn me once, shame on you. Burn be twice, shame on me.
So let's turn the page today. There were security issues with the Gizmo5 solution last week, and you can read all about it in last week's article if you're interested. Suffice it to say, it appears there was no business relationship between Google and Gizmo5 so it was a really bad idea to be providing your gMail credentials to Gizmo5 since they also are tied to Google Checkout and other Google services.
Aug. 4 Update: The solution outlined below works reliably. But it has a couple of limitations which have been alluded to in the comments. Specifically, there is not a simple way to terminate a call to either an answering machine or an unanswered call. In addition, two dedicated DIDs are required to get inbound and outbound calls working through Google Voice. Based upon user suggestions and a few new tricks of our own, we now have a tweaked solution that requires only a single DID and that properly terminates calls when either party hangs up. Please read the updated article here. Our apologies... recalling the Wild West adage: "Pioneers are the ones with the arrows in their backs."
Today's solution is much more secure. Your Google Voice credentials are stored on your Asterisk server and are only used to make HTTP requests to Google Voice to initiate calls. If you're just getting started with Google Voice, be aware that you do not have to use your gMail address to set up your Google Voice account, but it's a good idea particularly if you plan to use other Google services. For example, if you get an Android cellphone (as we did last week), the entire phone setup is configured around your gMail address and password. And, to integrate Google Voice into the phone, you need a gMail address that's tied to your Google Voice account. Once you make that link, all U.S. calls on your Android phone are free if they're routed out through Google Voice. Very slick!
Let's get down to business now. We promised you a secure Google Voice interface to Asterisk this week, and we've got it. As this is written, the only restrictions on use to make free U.S. calls are those set forth in the Google Voice Terms of Service. When we're finished today, you'll be able to pick up any Asterisk phone and dial GV-678-123-4567 to make a free call to anywhere in the United States and Canada. The call will go directly from your Asterisk server, through Google Voice, to the destination number you specify. In short, there is no intermediary in completing the calls. Google obviously can change the terms of service but, until they do, you should have a stable and free provider for making and receiving calls within the United States.
Special Thanks. At the outset, we want to express our sincere appreciation to Jacob Feisley and Paul Marks for their pioneering work on a Python interface to Google Voice. We also stumbled upon another Python development project, Google Voice for Python. While we originally had planned to rely upon Jacob and Paul's script, we ultimately decided to implement pygooglevoice because of the additional flexibility it provided for down the road. With pygooglevoice, you not only can make Google Voice calls, but you also can send SMS messages with no muss or fuss. Jacob Feisley has now joined that project as well. So, our special tip of the hat goes to the entire Google Voice for Python development team. It's a terrific product as you will see.
Prerequisites. Today's setup requires a CentOS-based Asterisk aggregation with a current version of FreePBX. Be aware that today's solution requires Python 2.4 and reportedly will not work with Python 2.5 found in some Linux distributions. We've tested everything with PBX in a Flash and, on that platform, you're good to go. The install script should work equally well with the other CentOS-based Asterisk aggregations, but we haven't tested them. Be our guest, and let us know if you encounter any problems. Finally, a word of caution. We don't ordinarily distribute solutions using development tools we don't use. Our knowledge of Python wouldn't fill a thimble. We've made an exception today because of the extraordinary interest in Google Voice by the Asterisk community. But, if something comes unglued, we can't fix it. So have a backup plan in place just in case. 🙂
Today's New Design. Here's the new design. You obviously still need a free Google Voice account. If you don't have one, you can request an invite here. At last report, it's only taking about a week from application to invite which is really great news! Once you have a Google Voice account and phone number (Google has reserved a million of them so... not to worry!), then we'll set up a couple of forwarding phone numbers in your Google Voice account. You aren't actually going to use one of them, but it does have to be registered as one of your GV forwarding numbers. We'll explain in a minute. Unlike the previous SIP approach, you no longer have to configure your Google Voice account to forward all incoming calls to voicemail. As you may recall, this allowed you to call your Google Voice number and press a few keys to make an outbound call instead of listening to your voicemails. The new Python approach simulates the Google Voice click-to-dial web interface to place calls so it won't disturb the normal use of your Google Voice phone number. But you need a RingBack number for the calls.
With a little dialplan voodoo and pygooglevoice, we'll tell Asterisk to place a click-to-dial call using one of your Google Voice forwarding numbers. Then we'll stuff in the destination U.S. phone number. When you dial GV-678-1234567 from any of your Asterisk phones, Asterisk will connect both your extension and the called party to an Asterisk MeetMe Conference where you can chat as long as you like. The entire procedure is totally transparent both to the caller and the callee. And the sound quality is still great.
To use Asterisk for inbound calls through Google Voice, you can designate a second forwarding number in Google Voice that connects to one or more extensions on your Asterisk system. All you really need for this is any DID, but you'll need separate DIDs to handle inbound and outbound calls. We'd recommend free IPkall and SIPgate DIDs. For outbound calls, we'll point one of your DIDs that's registered with Google Voice to a special Google Voice MeetMe conference line on your PBX. For inbound calls, point the other DID registered with Google Voice to an extension or ring group on your Asterisk server. Since all of these calls are free, the area code of the two DIDs really doesn't matter because you're never going to publish the fact that they exist.
Update: See the comments to this article for a one-DID solution that can handle both inbound and outbound calls.
To get everything working today, there are four steps: (1) configuring two Google Voice forwarding numbers to manage your outbound and inbound calls, (2) configuring FreePBX to route all outbound calls with a GV prefix to your special Google Voice dialplan context, (3) setting up a reserved MeetMe conference to manage your outbound Google Voice calls, and (4) running our install script which adds the dialplan code for Google Voice outbound dialing with your credentials and puts the Python application into place on your server. It sounds more complicated than it is. So hang on to your hat. Here we go!
Getting Two DIDs. Before you can use Google Voice with Asterisk, you'll need a couple of DIDs. As we indicated, we'll use one of the DIDs to handle the GV RingBack process when it places outbound calls. The other DID will be used to route incoming calls from Google Voice to your Asterisk server when people call your GV number. To get started, use the links above to obtain and configure the two DIDs. Then point each of them to an extension on your Asterisk system that can be used to verify your requests for the numbers.
The easiest method for setting these DIDs up is to create SIP URIs for the two DIDs on your Asterisk system. Then point each DID to a separate SIP URI on your server. Route each SIP URI to an Inbound Route in FreePBX where you can manage the destination for that DID's calls.
HINT: The entry in extensions_override_freepbx.conf would look something like this for a SIP URI called ipkall-1:
exten => ipkall-1,1,Goto(from-trunk,${DID},1)
Then you would create an inbound route named ipkall-1 using FreePBX and designate some existing extension on your server as the destination for these inbound calls.
When you set up the SIP forwarding for the DID at ipkall.com, you'd specify the SIP URI as:
ipkall-1@ipaddress_of_your-Asterisk_server
We've previously covered in detail how to do this so read the article if you need a refresher course. To reiterate, the area codes of these DIDs really don't matter because you're never going to give out either number. So use the free sources and save yourself some money. The real trick is you want to use DIDs with unlimited, free inbound calls. Both IPkall and SIPgate provide that functionality at no cost.
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. Add the area code and phone number of both DIDs: the RingBack DID for outbound calls and the Destination DID for incoming calls. Remember, you'll need to be sure both DIDs are pointed to extensions on your PBX that you can answer since you have to go through Google's confirmation drill to successfully register each number. After the RingBack DID is confirmed, be sure you uncheck this Google Voice destination. You do not want incoming Google Voice calls routed to this DID because it is going to land in a MeetMe conference on your Asterisk system. We're only setting it up as a destination number in Google Voice so that we can use it for click-to-dial ringbacks.
The RingBack DID will ring whenever you place a GV-routed Outbound Call from any phone on your Asterisk system. Remember, we're using click-to-dial technology to make these calls. You just won't see it. So, once this DID has been set up in Google Voice and you've confirmed that it works, we're going to alter the destination of this DID's inbound route and redirect it to a dedicated MeetMe conference number to manage these calls. Whenever you place an outbound call through Google Voice, pygooglevoice will initiate the click-to-dial process and then our dialplan code will transfer both the caller and the called party to the same dedicated MeetMe conference. When the other party answers, you both can begin talking. The MeetMe conference will be transparent to both parties except you'll get music on hold if either party hangs up.
Integrating Google Voice into Asterisk and FreePBX. This setup lets you place a call through Google Voice from any Asterisk phone by dialing the GV prefix plus a 10-digit number. So, to place a call to President Obama in Washington through Google Voice, you'd dial 48-202-456-1111.
First, open FreePBX with a web browser and choose Setup, Trunks, Add Custom Trunk. Insert the following Custom Dial String on the form and Submit Changes and reload the dialplan:
local/$OUTNUM$@custom-gv
Next, choose Setup, Outbound Routes, Add Route and fill in the following entries on the form:
Route Name: GoogleVoice
Dial Pattern: 48|NXXNXXXXXX
Trunk Seq: local/$OUTNUM$@custom-gv
Next, choose Admin, Setup, Conferences, Add Conference. For the conference number, make up an obscure 5-digit number. Name the conference GV. Leave both PINs blank. For Conference Options, use the following. Once you've completed all of the entries, click Submit Changes and then reload your dialplan.
- Join Message: None
- Leader Wait: No
- Quiet Mode: Yes
- User Count: No
- User join/leave: No
- Music on Hold: Yes
- Allow Menu: No
- Record Conference: No
Finally, for the inbound route of the DID that will handle the RingBacks, change the inbound route destination for this DID to the MeetMe conference number you created above.
Save your changes and reload the Asterisk dial plan one more time to complete the setup.
Integrating pygooglevoice. Now we're ready to complete the setup by running our little script which loads pygooglevoice and sets up your dialplan in extensions_custom.conf. You'll need 4 pieces of information to run the script so write them down before you begin:
1. Your Google Voice email address
2. Your Google Voice password
3. Your 11-digit RingBack DID (16781234567)
4. MeetMe Conference Number for Outbound Calls
A word of caution: If you used a gMail address to set up your Google Voice account, it's possible to have different gMail and Google Voice passwords. For this to work, you'll need to enter your gMail password, not your Google Voice password (assuming they're different).
Now log into your Asterisk server as root and issue the following commands:
cd /root
wget http://bestof.nerdvittles.com/applications/gv/install-gv
chmod +x install-gv
./install-gv
Google Voice Speed Dials. For frequently called numbers, you can add speed dials by inserting entries in the [from-internal-custom] context of extensions_custom.conf that look like the example below where 333 is the speed dial number and 6781234567 is the area code and number to call. Be sure to reload your Asterisk dialplan to activate them.
exten => 333,1,Dial(local/6781234567@custom-gv,300)
Congratulations! You now have free U.S. calling on your Asterisk system using Google Voice with no gimmicks and no strings. Enjoy!
If you happen to be running Asterisk 1.6, there's a simpler way to bridge your calls together without using the MeetMe conferencing trick. See this message thread in the PBX in a Flash Forum for details.
Finally, a word of caution. This transparent call design obviously only works for a single call at a time. There are no safeguards to prevent multiple calls, but that may violate the Google Voice terms of service anyway. And, even if Google Voice put the second call through, the end result would be that all four of you (2 callers + 2 destinations) would end up in the same conference. So much for privacy.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
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...
Nerd Nirvana: Free Google Voice Calling Returns to Asterisk
In what can only be described as a telephony game changer, Google Voice this past weekend expanded the scope of its offering by providing transparent SIP connectivity through Gizmo5 for inbound and outbound calling. Simply stated, you now can connect virtually any telephone to Google Voice using a garden-variety Internet connection. And the phone can be almost any SIP telephone or a standard home telephone plugged into a $40 ATA. Letting folks make click-to-dial calls through a PC is too geeky for most. But today's offering is a new animal. Google Voice now works with regular telephones.
Did we mention that you get a free phone number of your choice in almost any area code? Did we mention that every call you make throughout the United States and Canada is free? And, believe it or not, transparent Asterisk® support works out of the box as well. If your bread and butter business is SIP termination services in the United States (Are you listening, Vonage and Comcast?), then today probably isn't going to be your lucky day. For everyone else, it may just be remembered as the most important telephony development since the breakup of Ma Bell's monopoly. And now it's clear why Google Voice reserved a million DIDs. They're going to need every one of them... and more! Meet your New Phone Company®, Goliath Google, Inc. What Google Voice was missing was a simple interface to standard telephones, softphones, and SIP. Gizmo5 provides all of those missing pieces... and so much more. How about an almost-free Skype interface for openers.
As many of you know, we were ecstatic when Google Voice arrived with free U.S. calling, voice mail transcription, and SIP connectivity to Asterisk. Our solution lasted less than a week until Google slammed the SIP door and spoiled our party. So we shifted gears and showed you how to use a free Gizmo account and a free Google Voice account to make free SIP calls using Asterisk. Well, that lasted about a week as well although Craig Walker, who founded GrandCentral and now serves as the Google Voice Product Manager, responded to my inquiry about SIP support saying it sounded like a good idea and they would consider it once the initial Google Voice rollout was complete. Guess what? They've kept their promise.
Ironically, we had planned to introduce a new Google Voice solution for Asterisk today and were putting the finishing touches on the article when this news broke over the weekend. We've decided to postpone that discussion because, frankly, the Google Voice-Gizmo5 SIP marriage is the right way to go. It's straight-forward. It's proven technology. It's rock-solid reliable. And it's FREE!
Newly discovered issues with both security and Gizmo5's business model as pertains to making calls through Google Voice have given us pause in recommending the solution described below. In a nutshell, the solution below requires that you provide your Google email credentials to Gizmo5 in order to make the connection to Google Voice for free unlimited 20-minute 3-minute calling. Late yesterday, Gizmo5 announced a new 2¢ per minute fee for Google Voice calling (now described as Gizmo Voice). Yuck!
Even if you don't mind a stranger having unfettered access to your Gmail account, your Google credentials also may be used for other Google services including Google Checkout. Without a clearly defined business relationship between Google and Gizmo5, this would be a huge security risk. Having read several articles which hinted at a business relationship between Google and Gizmo5, we put our security concerns aside. However, when Gizmo5 began changing the ground rules for these calls (almost daily), it raised red flags that Google might not, in fact, be either a business partner or even a willing participant in Gizmo5's creation. As events continued to unfold, we have discovered that Gizmo5 may, in fact, be using a connection process that is not unlike the one we had planned to introduce this week anyway. And we have no business relationship with Google.
Bottom Line: Whether you are using an Asterisk server or not, WAIT! We have an equivalent, secure solution which is now available at no cost. We recommend you disable your Gizmo5-Google Voice setup if you already have put it in place and change your Gmail password! Then read the new Nerd Vittles article for a secure way to connect to Google Voice for free calling.
Our plan today is to show you the easy way to connect Asterisk to Google Voice through Gizmo5 to make free outbound phone calls and to receive free incoming calls. We'll leave the setup for a SIP phone, a generic Asterisk server, and an analog adapter such as the PAP2T-NA for another day. But we'll get to them sooner rather than later.
So, altogether now, welcome back... Googlified Messaging™. Before we begin...
Accounting 101. We hear you asking, "How long can the calls be free?" The short answer is probably not forever but long enough to run just about everyone else out of the business. Beyond that, what we see in our crystal ball pretty much lines up with Tim O'Reilly's talk at OSCON last week. And, at some point, Google may give you a choice of paying for the calls or perhaps volunteering to be their guinea pig for the mother of all indexing experiments. You'd agree to let them record your voice calls without identifying you individually. Then they could transcribe and index all of the keywords in your conversation and use those to identify buying trends, favorite movies, whatever. Remember, you can already say "Pizza" on your iPhone and get a list of nearby pizza parlors so this isn't as far-fetched as you may think. And keep in mind that, in some states, you only need the permission of one party to a telephone conversation to make a recording. Thanks to Amazon, it's been quite a resurgence for Big Brother. We thought we'd join the party with a little Orwellian hypothesizing of our own.
Step #1. If you're starting from scratch, the easiest way to get everything working today including Asterisk is to begin by installing PBX in a Flash, and then run the Orgasmatron Installer. This puts all the pieces in the proper places, and you'll be up and running in under an hour. For the complete soup-to-nuts tutorial, start here.
Step #2. You obviously still need a free Google Voice account to use Google Voice or Google Voice Dialing through Gizmo5. So that's next. If you don't have a Google Voice account, you can request an invite here. Our non-scientific survey suggests that it's taking less than a month to get an invite after you apply. YMMV! Once you have a Google Voice account and a local phone number (Google has reserved a million of them so... not to worry!), then you're all set.
Step #3. Next, you need a Gizmo5 account. If you don't have one, you can sign up for one within FreePBX once you run the Orgasmatron Installer. Or, you can download a Gizmo5 softphone and sign up that way. We're not sure it's required, but be charitable. Put a little money in your Gizmo5 Call Out account. You'll have it for a rainy day or international calling.
Step #4. We'll set up at least one forwarding phone number in your Google Voice account to match your Gizmo5 number. You don't have to actually use it, but it does have to be registered as one of your GV forwarding numbers. Unlike our previous SIP tutorials about Google Voice, you no longer have to configure your Google Voice account to forward all incoming calls to voicemail. As you may recall, this allowed you to call your Google Voice number and press a few keys to make an outbound call instead of listening to your voicemails. With the new Google Voice-Gizmo5 SIP offering, you no longer have to jump through all those hoops. It's a straight SIP-to-SIP-to-SIP connection from your Asterisk server to Gizmo5 to Google Voice.
Step #5. To use Asterisk for incoming calls through Google Voice, you can designate a forwarding number in Google Voice that connects to one or more extensions on your Asterisk system whenever anyone calls your Google Voice number. All you really need for this is one DID. This could be your Gizmo5 number, or it could be a free IPkall or SIPgate DID that's pointed to an extension or ring group on your Asterisk server. Since all of these calls are free, the area code of the DID really doesn't matter. The only number that will really matter to your callers is your main Google Voice number so be sure to select one for your hometown. Incidentally, you can add other forwarding numbers in Google Voice that will ring simultaneously with the DID on your Asterisk server. This could be your vacation home, your cell phone, or even your office phone.
Getting Started. We're going to be jumping back and forth between your Google Voice account, your Gizmo5 account, and the FreePBX web interface to your Asterisk server. So open each account in a separate tab with your web browser. To keep things simple, we're going to assume that you'll be using your Gizmo5 account to connect to your Asterisk server. In Asterisk lingo, the Gizmo5 account looks like any other DID on your Asterisk system.
FreePBX Setup for Gizmo5. If you've run the Orgasmatron Installer, you'll have a new Gizmo5 Integration option under the Setup tab. When you click on that option, you have the choice of either creating a new Gizmo5 account or using your existing account. Fill in the blanks to activate or create your new Gizmo5 account.
Once you've logged in, click Gizmo5 Integration Main Page. Choose Send all calls (except local extensions) through Gizmo5 and click Update Outbound Routes. For the time being, make certain that you have a default inbound route that rings one or more functioning extensions on your Asterisk system. You have to be able to answer an incoming call to complete the next steps. Finally, click on the Outbound Routes option. In the far right column, move the Gizmo5 entry to the top of the list and reload your dialplan when prompted.
If you're using a FreePBX-based system that doesn't have the Gizmo5 Integration option, you'll first need to establish an account at Gizmo5.com by downloading one of the softphones and signing up. After you have completed the sign up process, be sure that you disable automatic startup of the softphone. You can't have your Asterisk system AND the softphone registering to the same Gizmo5 account!
Next, using FreePBX, Add a new Trunk named Gizmo5. For the Peer Details, insert the following using your actual Gizmo5 phone number and password:
type=peer
insecure=very
host=proxy01.sipphone.com
username=1747XXXXXXX
fromuser=1747XXXXXXX
fromdomain=proxy01.sipphone.com
secret=password
context=from-gizmo5-trunk
qualify=yes
Leave the Incoming Settings section blank and then enter the Registration String using your actual Gizmo5 phone number and password:
1747XXXXXXX:password@proxy01.sipphone.com
Save your settings and reload your dialplan when prompted.
Next, create a Default Inbound Route so that calls from Google Voice will be routed to extensions on your server. Then, create an Outbound Route called OutGizmo with NXXNXXXXXX and 1NXXNXXXXXX as the Dial Patterns and Gizmo5 as the main Trunk Sequence . Move this route to the top of your outbound routes to assure that U.S. calls are placed using the Gizmo5 trunk. Reload your dialplan when prompted.
Finally, log into your Asterisk server as root and insert the following lines at the end of extensions_custom.conf in the /etc/asterisk directory. Then reload the dialplan: asterisk -rx "dialplan reload"
[from-gizmo5-trunk]
exten => s,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,Goto(from-trunk,${DID_EXTEN},1)
Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. This forwarding phone number should be the DID that you want Google Voice to call when you have incoming calls on your Google Voice number. Again, to keep things simple, add your Gizmo5 phone number (747XXXXXXX) and select Gizmo as the Phone Type. You then will be prompted to place a test call and provide a 2-digit number to verify that the number is working. Answer the extension on your Asterisk system when it rings and enter the 2-digit code that's provided.
Gizmo5 Configuration. Log in to your Gizmo5 account using your 1747XXXXXXX account number or username and password. In the new Google Voice section of the form, insert your Google Voice email address and password. This is the email address you used to set up your Google Voice account. Choose "Use for U.S. calls only" and then click SAVE.
July 29 Update. Since this article was released, Gizmo5 has reduced the allowable calling time from unlimited to 20 minutes. Then today it was reduced to 3 minutes. That may be as long as you like to talk on the phone, but it's a major change from what was initially introduced 3 short days ago. Looks like we'll dust off our original article after all. Stay tuned...
Deals of the Week. The nation's premier provider of free directory assistance service, 1-800-FREE-411, now is offering free 5-minute phone calls to most destinations around the world. Just listen to two quick commercials and enjoy your free call. Thanks, @MichiganTelephone. And now you can send free SMS messages worldwide from your iPhone. Thanks, @TruVoIP. Finally, AT&T has the refurbished 8GB iPhone 3G for $49 with a two-year contract.
Originally published: July 26, 2009
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you're wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what's happening. It's a terrific resource both for us and for you.
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...
VoIP Over VPN: Securely Interconnecting Asterisk Servers
We’ve just returned from a week in the Pacific Northwest teaching an Asterisk® course for an organization that wants to interconnect satellite offices using Asterisk servers. This coincided with a support request from one of America’s premier airlines which wants to do much the same thing for all of its reservation counters in airports situated in feeder cities around the country. Suffice it to say, PBX in a Flash in conjunction with Asterisk and Hamachi VPNs is perfectly suited to let anyone build these interconnected systems in minutes rather than months. In fact, with less than a day’s worth of introduction to Asterisk and PBX in a Flash, a group of 16 network administrators with no previous Asterisk experience did just that in a one-hour lab session during our training seminar last week. At the risk of (further) destroying our ability to earn a living, here’s how we did it.
Proxmox as a Training Tool. Before we get into the nitty gritty of actually interconnecting Asterisk servers with Hamachi VPNs, let us provide the free tip of the week for those of you that want to experiment with interconnecting Asterisk servers or for those that like to test various Asterisk scenarios without rebuilding servers all day long. There is no finer tool for this than the Proxmox Virtual Environment, a free and easy to use Open Source virtualization platform for running Virtual Appliances and Virtual Machines. With a sale-priced Dell T105 with a Quad Core AMD Opteron processor and 8 gigs of RAM, you’ll have a perfect platform to run about 16 simultaneous PBX in a Flash servers. The trick is finding the machines on sale for half price which is about every other week. Our lab system which matches this configuration was less than $600 with RAM purchased from a third party. You can save most of the shipping cost by using our coupon link in the right column to shop at Dell’s small business site.
Proxmox lets you build virtual machines in two ways: OpenVZ templates or Qemu/KVM Templates and ISO images. While we intend to offer an OpenVZ template for PBX in a Flash soon, currently it’s easy to create your own ISO template using the standard PBX in a Flash ISO image. Once you’ve uploaded your ISO image into Proxmox, simply create a new virtual machine by giving it a name, specifying 512MB of RAM and a 30GB partition. In 10 seconds or less, your new VM will be ready to boot. Start your VM and then open the VNC console window within the Proxmox web interface and install PBX in a Flash just as if you were building a stand-alone machine. When the 15-minute install completes, run through the Orgasmatron Installer setup, and you’ll have your turnkey PBX in a Flash system ready for production in less than 30 minutes.
You don’t have to repeat this drill for every virtual machine. Instead, use the built-in Proxmox backup utility to make a backup image of what you built. Shut down the VM, create a /backup directory, and then schedule the compressed backup in the web browser. When the backup completes, you’ll have a backup image in /backup with a file name like this: vzdump-101.tgz.
To create a new virtual machine, you issue the following command while positioned in the /backup directory specifying the number for the new virtual machine:
vzdump --restore vzdump-101.tgz 102
In about 3 minutes, you’ll have a second virtual machine that’s a clone of the first one. Because it’s a true clone, it would obviously have the same MAC address for the virtual NIC. You don’t want that or all of your VMs would boot up using the same IP address. Using the Proxmox web interface, just edit the new VM 102 by switching from the Status tab to the Hardware tab, delete the existing Ethernet device, and then create a new Ethernet device under the hardware address list pulldown. This will create a new virtual NIC with a new MAC address. So, when you boot VM 102, it will be assigned a new IP address by your DHCP server. You can decipher the new IP address by opening the VNC console window for VM 102 after you boot it up. Now you’re an expert. You can create the additional Baker’s Dozen turnkey PBX in a Flash servers in about an hour. Start all of them up, and you’ve got an instant training facility and PBX in a Flash playground.
April, 2012 Update. See our new article for a current state-of-the-art VoIP VPN.
Creating Hamachi VPN. You obviously don’t need a virtual private network in order to interconnect Asterisk servers. But, as easy as the Hamachi VPN is to set up, especially with PBX in a Flash servers, why wouldn’t you want all of your inter-Asterisk communications secured and encrypted? In addition to the capacity limitation of the Proxmox server, there’s another reason we chose to build 16 PBX in a Flash VMs. That happens to be the number of servers you can interconnect with the Hamachi Virtual Private Network without incurring a charge.1 Why use the Hamachi VPN when OpenVPN is free with unlimited network connections and no strings? The short answer is it’s incredibly simple to set up without public and private key hassles, and it supports dynamic IP server addressing with zero configuration. We plan to cover OpenVPN in a subsequent article but, for many implementations, Hamachi VPNs offer a robust, flexible alternative that can be deployed in minutes.
If you’re not using PBX in a Flash, there are a million good Hamachi VPN tutorials available through a quick Google search. If you are using PBX in a Flash, we’ve done the work for you. With the Orgasmatron Installer build, you’ll find the Hamachi VPN installation script in /root/nv. For other PBX in a Flash systems, just download the install-hamachi.x script from here or, after logging into your server as root, issue the following commands:
wget http://pbxinaflash.net/source/hamachi/install-hamachi.x
chmod +x install-hamachi.x
./install-hamachi.x
Before beginning the Hamachi VPN install, it’s a good idea to make yourself a cheat sheet for the servers you plan to interconnect. We’re going to interconnect 3 servers today, but doing 16 is just more of the same. You’ll need a unique name for your virtual private network. Pick a name that distinguishes this VPN from others you may build down the road. For our example, we’re going to use piaf-vpn. Next, you need a very secure password for your VPN. We’re going to use password for demonstration purposes only. Finally, you need a unique nickname for each of your servers, e.g. piaf-server1, piaf-server2, and piaf-server3 for our example setup today.
For the first Hamachi install, we’ll need to create the new network. For the remaining installs, we’ll simply join the existing network. Keep in mind that you can only remove machines from the network using the same server that was used to create the other VPN accounts initially so build out your virtual private network by starting with your main server, piaf-server1 in our example.
To begin the Hamachi VPN install, run the script using the commands shown above. Type Y to agree to the installer license and then press the Enter key to kick off the install. For the piaf-server1 install, type N to create a new Hamachi network. For the remaining installs, you’d type J to join an existing Hamachi network. Enter the network name you chose above. For our sample, we used piaf-vpn. Type it twice when prompted. Now type your network password and then your nickname for this server when prompted to do so. Then standby while the Hamachi software is installed. It takes a few minutes depending upon the speed of your network connection. And remember, do NOT use our sample network name. Make up your own and don’t forget it. When the install completes, you can review the log if you’d like. Unless something has come unglued, Hamachi should now be running on your first server. Repeat the drill on your other servers.
The next step is to grab some of our scripts to make it easier to manage Hamachi on your servers.
cd /usr/local/bin
wget http://pbxinaflash.net/source/hamachi/hampiaf
wget http://pbxinaflash.net/source/hamachi/hamachi-servers
chmod +x ham*
cd /root
wget http://pbxinaflash.net/source/hamachi/hamachi.faq
The hamachi.faq document provides all of the commands you’ll need to manage Hamachi including the steps to start over with a totally new virtual private network. For now, let’s be sure your network is running. Type: hamachi-servers piaf-vpn using the network name you assigned to your own VPN. Then type it again, and it should display all of the servers on your VPN with their private VPN IP addresses:
root@pbx:~ $ hamachi-servers piaf-vpn
This server:
Identity 5.151.123.1
Nickname piaf-server1
AutoLogin yes
OnlineNet piaf-vpnGoing online in piaf-vpn .. failed, already online
Retrieving peers’ nicknames ..
* [piaf-vpn]
5.151.123.2 piaf-server2
5.151.123.3 piaf-server3
Finally, a word of caution about security. One of the drawbacks of the ease with which you can create Hamachi VPNs is the ease with which you can create Hamachi VPNs. Anyone that knows your network name and password can join your network with one simple command. You can kick them off from the main server where the VPN was created (hampiaf evict piaf-vpn 5.249.146.66), but you can’t keep them from joining. So, protect your network by making the password extremely secure. There currently is no way to change your network password. All you can do is create a new network with a new network name and a more secure password.
Interconnecting Asterisk Servers. Once your VPN is established and all of your servers are on line, then we’re ready to interconnect them with Asterisk and FreePBX. There are a number of ways to do this. For smaller networks, we’re going to show you the easy and secure way using IAX and the VPN you just created. As with the VPN setup, a cheat sheet comes in handy to avoid erroneous entries that would cause your calls between servers to fail. What we recommend is assigning and creating a block of extensions on each of your servers with different ranges of numbers. For example, we’re going to use four-digit extensions in the 1xxx range for piaf-server1, 2xxx for piaf-server2, and 3xxx for piaf-server3. The idea here is that the extensions are unique between your servers. This makes it easy to dial between offices without having to resort to dialing prefixes. So the first step in interconnecting your servers is to build the necessary extensions on each of your servers.
Now for the cheat sheet. Using the hamachi-servers tool above, decipher the VPN IP address of each of your servers and make a chart with the server names, the range of extension numbers, and the VPN IP address of each server. You’ll also need to think up a very secure password. We’re going to use the same one for all of the servers although you certainly don’t need to. So long as the password you choose is secure, there’s really no reason not to use the same one.
piaf-server1 1xxx 5.151.123.1 password
piaf-server2 2xxx 5.151.123.2 password
piaf-server3 3xxx 5.151.123.3 password
Creating Trunks. The next step is to create an IAX trunk on each server for each remaining server in your network. In our example, on piaf-server1, we’d want to create trunks for piaf-server2 and piaf-server3. On piaf-server2, we’d want to create trunks for piaf-server1 and piaf-server3. And so on.
NOTE: Because of a change in IAX design to fix a security issue that arose after this article was originally published, be sure to add the following line in the User Details of each trunk below:
requirecalltoken=no
On your first server (piaf-server1 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server2 with the following entries. Leave everything blank except the entries shown below:
While still on piaf-server1, repeat the process to create a trunk for piaf-server3:
On your second server (piaf-server2 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server1 with the following entries. Leave everything blank except the entries shown below:
While still on piaf-server2, repeat the process to create a trunk for piaf-server3:
On your third server (piaf-server3 in our example), using a web browser, open FreePBX and choose Admin, Setup, Trunks and then click Add IAX2 Trunk. Create the trunk to piaf-server1 with the following entries. Leave everything blank except the entries shown below:
While still on piaf-server3, repeat the process to create a trunk for piaf-server2:
Creating Outbound Routes. Now we need to tell Asterisk how to route the calls between the servers. In a nutshell, we want calls to extensions in the 1xxx range routed to extensions on piaf-server1, calls to 2xxx extensions routed to piaf-server2, and calls to 3xxx extensions routed to piaf-server3. On each server, create an outbound route for each of the remaining servers. Name the routes server1, server2, and server3 as appropriate. The critical pieces of information in each outbound route are the dial string (which should match the extensions on the server we’re connecting to) and the Trunk Sequence (which should be the appropriate IAX trunk for the server we’re connecting to).
On piaf-server1, we’d have a server2 outbound route with a Dial String of 2xxx and a Trunk Sequence of IAX2/piaf-server2. Then we’d have another server3 route with a Dial String of 3xxx and a Trunk Sequence of IAX2/piaf-server3. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.
On piaf-server2, we’d have a server1 outbound route with a Dial String of 1xxx and a Trunk Sequence of IAX2/piaf-server1. Then we’d have another server3 route with a Dial String of 3xxx and a Trunk Sequence of IAX2/piaf-server3. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.
On piaf-server3, we’d have a server1 outbound route with a Dial String of 1xxx and a Trunk Sequence of IAX2/piaf-server1. Then we’d have another server2 route with a Dial String of 2xxx and a Trunk Sequence of IAX2/piaf-server2. If you have a catch-all outbound route, be sure to move these routes above the catch-all in the right column. Then reload your dialplan.
If you’re setting this up with PRI or T1 connections between your servers, you might also want to specify at least secondary trunk sequences for each of the outbound routes to provide some redundancy. For example, on piaf-server1, you might want a secondary Trunk Sequence for server2 that specified IAX2/piaf-server3. Then, if the primary connection between server1 and server2 was down, Asterisk would attempt to complete calls to 2xxx extensions by routing them to server3 and then on to server2 from there. To the caller and call recipient, they’d never know that the direct link between server1 and server2 had failed.
Alternate routing might also be appropriate where you have more capacity between certain servers. For example, if you had a single T1 line between server1 and server3 but you had PRI connections between server1 and server2 and between server2 and server3, then it might make more sense to indirectly route 3xxx calls from server1 through server2 and then on to server3 rather than the direct route from server1 to server3. Enjoy!
Free DIDs While They Last. Sipgate is giving away a free U.S. DID with free incoming calls plus 200 free minutes for outbound calls. Better hurry. Here’s the trunk setup for FreePBX-based systems:
Trunk name: sipgate
type=peer
username=ACCTNO
fromuser=ACCTNO
secret=ACCTPW
context=from-trunk
host=sipgate.com
fromdomain=sipgate.com
insecure=very
caninvite=no
canreinvite=no
nat=no
disallow=all
allow=ulaw&alawRegistration Strong: ACCTNO:ACCTPW@sipgate.com/YOUR-DID-NUMBER
ACCTNO is the account number assigned to your sipgate account. ACCTPW is the password for your account. YOUR-DID-NUMBER is your 10-digit DID.
Finally create an inbound route using your actual 10-digit DID and assign a destination for the inbound calls.
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
Twitter Magic. If you haven’t noticed the right margin of Nerd Vittles lately, we’ve added a new link to our Twitter feed. If you explore a little, you’ll discover that the user interface now brings you instant access to every Twitter feed from the convenience of the Nerd Vittles desktop. Enjoy!
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
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 comment #1 below. [↩]
Remotely Managing Your Asterisk Server with WebDAV
It's been quite a while since we last explored WebDAV, and that was in the context of turning a Mac into an ISP-in-a-Box in 2005. Today we want to do much the same thing with your PBX in a Flash server, and the drill is quite similar. Over the course of the last four years, the uses of WebDAV have grown geometrically.
Overview. As you probably know, WebDAV is an acronym for Web-based Distributed Authoring and Versioning. Simply put, it is an HTTP protocol extension that allows people anywhere on the Internet to collaboratively edit and manage documents and other files using the same protocol and port used for surfing the web. In the Mac world, WebDAV provides a Disk Volume on your Desktop that “looks and feels” like any other networked hard disk. In the Windows world, WebDAV is called Web Folders. They can be used like any other mapped drive in Network Neighborhood. If you’re still a little fuzzy about the WebDAV concept, think of how you link to another drive on your local area network. WebDAV gives you the same functionality across the entire Internet with virtually the same ease of use. Depending upon user privileges, of course, you can copy files to and from a WebDAV volume, and the protocol imposes versioning control through file locking to assure that multiple people with access rights don’t change the same file at the same time.
For openers, WebDAV provides a simple vehicle to manage your PBX in a Flash web site by letting you create a file-sharing link to your server which is read and write-accessible (with a password) from almost anywhere. It also could be used to upload and/or download sensitive corporate data, or it could serve as a backup repository for your portable or desktop PC. Think of it as a Poor Man's Cloud Computing alternative. Install a couple of terabyte drives on a Dell T100 or SC440, and you've got a secure environment for storing all of your data on a single server.
Initial Setup of WebDAV. For today, we're assuming you already have a functioning PBX in a Flash server. It includes most of the WebDAV components necessary to get WebDAV working. If you're using some other Asterisk® platform, then take a look at our previous articles for some hints on the basic setup keeping in mind that most Asterisk distributions use asterisk as the web user account rather than apache. To keep things simple, we're going to set up a separate dav directory within your existing PBX in a Flash web server to use for WebDAV access. This means files and folders managed with WebDAV will appear in /var/www/html/dav on your server.
To complete the WebDAV setup on PBX in a Flash systems, log into your server as root and issue the following commands:
mkdir /var/www/html/dav
chown asterisk:asterisk /var/www/html/dav
chown asterisk:asterisk /var/lib/dav
cd /etc/pbx/httpdconf
wget http://pbxinaflash.net/source/webdav/dav.conf
apachectl restart
Configuring WebDAV. As installed, you'll need your username (maint) and your password to access your WebDAV server from either a browser (for read access) or via network access (for read and write access). You have several choices in how to reconfigure this setup to meet your own requirements. If you want to upload and manage files in this directory with a password and then allow anyone to access the files with a web browser with no password, you can simply uncomment the two Limit lines in the Apache dav.conf file in /etc/pbx/httpdconf. Just remove the leading # characters from both the lines in the configuration file. If you want to restrict network and web access to WebDAV to certain IP addresses, you can remove the Allow from all line in dav.conf and add lines that look like the following:
Allow from 192.168.0
Allow from 68.218.222.170
Remember to give yourself access on the private LAN as well as the public side if you plan to use WebDAV from outside your firewall. Our strongest recommendation remains to not expose your server to public web access without restricting access with either passwords or IP restrictions in .htaccess files for each directory as shown above.
Accessing WebDAV. To access your WebDAV folder with a browser for read-only access, point your browser to the IP address of your server and then the /dav subdirectory. For example, on your private LAN, the link might look like this: http://192.168.0.123/dav. On the public Internet, the link might look like this: http://pbx.dyndns.org/dav.
On a Windows machine, you can create a Web Folder for access to your new WebDAV directory like this:
My Network Places
Add a network place
Choose network location
http://192.168.0.123/dav (with no trailing slash!)
username: maint password: yoursecretpassword
Name the link: PiaF WebDAV
Update: There is a glitch with Web Folder access with some Windows XP and Vista systems. Here's a link to the Microsoft Patch that addressed the problem.
On a Mac, click on your Desktop to open Finder and do the following:
Go
Connect to Server
http://192.168.0.123/dav (with no trailing slash!)
Connect
username: maint password: yoursecretpassword
OK
Using WebDAV for Total Web Site Management. There may be some who actually want to use WebDAV to manage your entire PBX in a Flash web site. This means all directories from /var/www/html down. This WebDAV management need not be exclusive. In other words, you still can retain the WebDAV setup for the dav directory outlined above. To add an additional WebDAV service for your entire web site, you will need to edit /etc/httpd/conf/httpd.conf. Then search for this text:
<Directory "/var/www/html">
Once you find that line of code, scroll down to just above the </Directory> entry and insert the following lines of code. Save your additions and then restart Apache: apachectl restart.
DAV on
<Limit PUT POST DELETE PROPFIND PROPPATCH MKCOL COPY MOVE LOCK UNLOCK>
AuthType Basic
AuthName "WebDAV Web Server Access"
AuthUserFile /usr/local/apache/passwd/wwwpasswd
Require valid-user
Order allow,deny
Allow from all
</Limit>
If you haven't also implemented the dav solution above, then make certain you issue the following command while logged into your server as root:
chown asterisk:asterisk /var/lib/dav
Now that you have your own WebDAV server, take a look at this terrific web site for some great ideas on what's possible in the open source and commercial world of WebDAV. Enjoy!
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