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: parkedcalls

Destination 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.

---

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

---

Some Recent Nerd Vittles Articles of Interest...

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-vpn

Going 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&alaw

Registration 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.

---

 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 

---

Some Recent Nerd Vittles Articles of Interest…

1. See comment #1 below. [↩]

We’ve previously written about the incredible Sonos whole-house audio system that is priced (literally) tens of thousands of dollars below the cost of a comparable "turnkey" system that you typically would purchase from a home audio consultant. Another revolutionary development occurred yesterday so it was a good time for an update.

Yesterday’s development was an announcement from Napster, which was recently acquired by Best Buy, that lets you download 5 DRM-free songs per month from Napster’s entire catalog for $5 a month. Nothing very exciting there. The kicker is that, for no additional fee, you now get unlimited (but DRM’d) streaming of all 7 million songs in Napster’s vast music collection to any PC you happen to own. And $60 buys you a full year plus 70 DRM-free songs!

We hear you mumbling. Why would anyone want to only listen to music on their PC? Well, this is where your Sonos music system comes into play. Instead of buying a cheap PC (such as this $199 Acer netbook from CompUSA) and subscribing to Napster to play the music on your PC, U.S. customers now have instant access on your Sonos system to over 7 million music tracks in the Napster library any time you like. And this isn’t canned playlists although Napster has plenty of those. With today’s new offer, you can stream songs of your choice in your own playlists to one or many rooms in your house depending upon how many Sonos ZonePlayers you’ve configured. Or use your Sonos controller to search the entire Napster catalog by artist, album, or song title. And the total cost: just $5 a month.

Sonos Background. For those that are new to Sonos, you basically buy a little $500 Wi-Fi box for each room in your home or office where you want to play music. There are special system bundles at this link if you hurry. You plug in a pair of speakers and connect to your NAS-savvy music library. We recommend dLink’s DNS-323 which provides RAID1 mirrored SATA drives in any size you desire (about $180 delivered from NewEgg plus SATA drives). Be sure the drives you pick are on dLink’s compatibility list! If you happen to use Comcast for your broadband service, you also receive a free Rhapsody subscription which can be played on every Sonos system in your house for free, but you’ll have to connect a Windows PC to your Sonos system through the line in jack to take advantage of this. With the new Napster offering, you can skip the hassle for $5 a month. The Sonos system also supports streaming audio from more than 300 Internet radio stations, also free.

Some other reviews of the Sonos system are worth a look. Check out the Home Theater View, Audioholics, Playlist Magazine, and PC Magazine. You’ll find dozens more here.

There are few companies in the world (much less the United States) that provide flawless hardware and software, free software updates (that always work), and regular updates that consistently add value to your initial purchase. Sonos is at the top of that very, very short list. Run, don’t walk, to add this system to your home or office. You’ll thank us for years to come. We installed eight systems with four remotes in just over two hours. We haven’t quit listening since. Today’s Napster announcement is simply icing on the cake. Enjoy!

Update. We don’t often revise our articles but a Tweet from @Sonos last night sent us back to the drawing board. While we knew that Napster already was available in Sonos music players, the price point was substantially higher. Since Napster’s announcement had clearly stated that the $5 a month special only applied to use of the library on a PC, we had assumed that it wouldn’t work directly in the Sonos system. Wrong! It works perfectly on the Sonos players with the functional simplicity that is the hallmark of Sonos software. Napster should take a lesson! Lo and behold, it appears that Napster views the Sonos system as just another Linux PC so the entire Napster music library is available in any Sonos music system without resorting to any external PC. Seven million songs for $5 a month strikes us as a deal you’d be crazy to pass up. Better hurry while it lasts.

---

Some Recent Nerd Vittles Articles of Interest…

PBX in a Flash offers a number of Asterisk- compatible PBX solutions to meet virtually every need. These range from base installs of Asterisk 1.4 and 1.6 in both 32-bit and 64-bit flavors. In addition, the Orgasmatron builds provide turnkey installs for Everex gPC systems and Dell PowerEdge SC440 and T100 servers. And our recent VPN in a Flash build for the Acer Aspire One NetBook introduced the ultimate portable, secure traveling communications server including the Hamachi VPN.

For 2009 we round out our offerings with the ultimate development tool, a bootable USB flash drive which can create turnkey, full-featured Asterisk PBX systems in 15 minutes or less. As its name suggests, this build was specially engineered for the new Atom-based motherboards found in most netbooks although it works just fine with Dell’s PowerEdge T100 servers as well. Many of the newer netbooks lack a CD/DVD drive so a bootable flash installer is ideal. In addition to a current generation computer, you’ll also need an 80GB or larger SATA disk drive which can be configured as sda1, sda2, and sda3. RAID setups are not yet supported unless you’re very familiar with reconfiguring Mondo Restores. With your new computer in hand, just plug in the Atomic Flash, and boot the computer from the flash drive. Type nuke and have a cup of coffee. When you return in 15 minutes and type a couple commands, your system will be ready for deployment. Add your trunk providers, match phones to the preconfigured extensions, secure passwords, and you’re all set. It’s that easy!

Make no mistake. This is a Bleeding Edge installer featuring a Fedora 10 Remix¹ that’s less than a week old. It supports the latest and greatest motherboards, wired and WiFi networks, and it includes the KDE graphical user interface for those that love GUIs. Out of the box, it provides a functioning softphone as well as your own private Hamachi VPN connecting up to 15 additional systems so the entire setup can be deployed as a mobile communications hub in less time and for less money than most folks spend on their breakfast.

For those that demo systems for a living, no one will touch this presentation. Just show up at a customer site with a $300 Acer Aspire One NetBook and an Aastra 57i business phone. While the customer watches the Atomic Flash build a new PBX in a Flash server from the ruins of a Windows XP clunker, you can connect and configure the 57i and explain how simple VoIP networks can be.

When you finish your 10-minute slide show, your system will be operational. Dial any 800 number from your Aastra phone, and presto… instant, flawless communications! Now explain to the customer what the world of penny-a-minute communications is all about with every call between PBX in a Flash systems and other SIP phones absolutely free… worldwide.

Friends of PIAF. So how do you get one? If you don’t mind a preproduction version, which means we have to custom-build every flash drive, here’s how to get yours. First, this offer is for a limited time (until we get sick of cloning flash drives). And don’t expect to receive your unit overnight. In fact, it may be several weeks or more depending upon how busy we get with other Honey-Do’s. But we won’t forget you!

Now what? Just make a contribution of $50 or more to the PBX in a Flash project through PayPal, and we’ll give you one (as in gift for free), and we’ll even pay the shipping. Limit of one per contributor please! Keep in mind that $50 barely covers the cost of the 8GB flash drive, the shipping, the PayPal commission, and the labor (at 5¢ an hour) so your generosity is most appreciated. And when we get tired of working for 5¢ an hour, we’ll holler.

Once your Atomic Flash device arrives, please visit http://atomicflash.org or http://pbxonaflash.com for complete installation instructions.

The Perfect Complement. The stars have all lined up to provide a perfect opportunity for you to purchase a state-of-the-art NetBook. Click or hover on the image above for details. If you’d prefer a server, you now can grab a Dell Poweredge T100 server with dual 160GB SATA drives and 2GB of RAM saving $397 off the list price. Either hardware works great with Atomic Flash.

Are You Crazy? Why Are You Doing This? Well, yes and because it’s the First Anniversary of PBX in a Flash! We want everyone to experience PBX in a Flash in all its greatness now that we’ve got it down to a 15-minute walk in the park. These are tough economic times for many businesses around the world, and we want you to help us spread the word about the savings that can be realized through Voice Over IP. We also want to encourage those of you on the fence about a career to enter the Asterisk® reseller community, and we’re doing our part by providing the perfect sales and development tool.

So now’s your chance. We hope you’ll tell every business acquaintance and friend you have about PBX in a Flash. And you have our heartfelt thanks for your continuing support. It’s been a blast!

---

Some Recent Nerd Vittles Articles of Interest…

1. Fedora and the Infinity design logo are trademarks of Red Hat, Inc. Asterisk is a registered trademark of Digium, Inc. All other trademarks and registered trademarks are property of their respective owners. This software aggregation is neither provided nor supported by the Fedora Project and contains non-Fedora and modified Fedora content. Official Fedora software is available through the Fedora Project website [↩]

We’ve never quite forgiven Apple¹ for bricking some of the original iPhones because some owners chose to jailbreak their private property to learn how it worked or to add additional functionality. It may turn out to be Steve Jobs’ billion dollar blunder! The stunt was especially egregious when one considers that both the iPhone and much of Mac OS X are based upon open source software for which Apple didn’t pay a nickel. Apple certainly added a pretty wrapper, but the internals of both the iPhone and Mac OS X contain loads of pure open source code including dozens of Mach 3.0 and FreeBSD 5 applications. Destroying people’s cellular phones for accessing soft- ware that was licensed to Apple as open source code just doesn’t pass the smell test.

Thus it was with mixed emotions that we unwrapped our Apple TV during Christmas 2007. Like the iPhone, it was locked up tighter than a drum even though the internals of the product read like a Who’s Who of the Open Source Movement: awk, bzip, cut, grep, find, ftp, finger, gzip, more, nano, openssl, perl, sed, tail, tar, touch, uname, whois, zip, and on and on. In fact, Mac OS X arguably is a better Linux than Linux. Suffice it to say, we read numerous articles outlining the lengths to which some talented users were going to unlock their Apple TVs. The process required disassembly of the unit, removal of the hard disk, and then a tedious unlocking scenario that was akin to breaking into Fort Knox. We chose to leave our Apple TV in its shrink wrap.

So what’s wrong with the Apple TV? Well, nothing… if you don’t mind paying Apple over and over again to reacquire media content which you already have licensed and if you don’t mind jumping through the iTunes hoops to transfer that content to a device which is perfectly capable of being self-sufficient. Let’s see. $1.99 to watch a TV show or play a music video that’s already sitting on your TIVO machine or that’s already freely (and legally) available from numerous sources on the Internet. Apple has added YouTube access, but the design really limits you to the most popular content. That makes it unsuitable (or worse) for anyone under the age of 13… or over the age of about 25.

Fast forward to 2009, and we decided it was time to take another look at the Apple TV landscape. WOW! What a difference a year makes. You now can create a bootable USB flash drive in a couple minutes, plug it into your Apple TV, and have a perfectly functioning, (true) open source appliance with DIVX and AVI support in less than 15 minutes. The FrontRow-enhanced Apple TV provides access to virtually all media content in every format imaginable with incredibly slick user interfaces thanks to the XBMC Media Center, Boxee Social Media Center, Nito TV, and Hulu. Most were originally designed for Microsoft’s Xbox. Uploads and downloads of media content can be performed using either your Apple TV controller and a television, or a web browser, or SAMBA networking, or SSH. So thanks to a resourceful bunch of talented, open source developers, we finally have an Apple TV worth owning that also happens to be fun to use. Incidentally, this whole metamorphosis can be accomplished without damaging the Apple TV’s existing user interface or its out-of-the-box functionality… at least until the next update from Apple.
So proceed at your own risk!

Freeing Your Apple TV. Since October, 2008, the emancipation of the Apple TV has become a simple, 5-minute exercise. What you’ll need to get started is an Apple TV² with version 2 software, a 1GB USB Flash Drive, and ATVUSB-creator which is free. The drill here is to create a bootable flash drive that can be used to reboot the Apple TV and transform its closed and proprietary shell into an open source platform. The preferred machine for creating your bootable flash drive is a Mac running Tiger or Leopard although a Windows XP/Vista solution is also available now. The only precaution we would add is to unplug all of the USB drives connected to your PC before creating the bootable flash drive. Then you won’t accidentally reformat the wrong USB drive. The one-minute CNET tutorial is here. A better one is here.

Once you have your bootable USB flash drive in hand, unplug your Apple TV and plug the USB drive into the unit. Now connect your Apple TV to a television. Power up your Apple TV and marvel at the installation process which takes under a minute. Whatever you do, don’t boot your Apple TV with the flash drive more than once! When the install completes, you should see a message indicating that your Apple TV can be accessed with SSH within a few minutes at frontrow@appletv.local. The password is frontrow. The IP address for your Apple TV also can be used for SSH access as well. Remove the flash drive and reboot. You’ll see a new menu option for XBMC/Boxee. Just follow the menu items to install both applications. After another reboot, you’ll be all set. Click on the CNET video above to watch a demo.

After installing the apps, launch and then configure XBMC. If you get an error that reads "Cannot launch XBMC/Boxee from path," it means you forgot to install the software through your TV menu. If you enable the web interface, you’ll be able to go to any browser on your LAN and manage XBMC through the following link using the IP address of your Apple TV: http://192.168.0.180:8080. For complete documentation, check out the XBMC Wiki.

Before you can use Boxee, you’ll need to visit their web site and sign up for an account. A tutorial on the application is available at UberGizmo. As luck would have it, this application only became publicly available in Alpha last week so we’re just in time. Don’t sweat the Alpha status too much, it previously ran on the XBox platform as well as Windows, Macs, and Linux. There’s social networking support via Twitter, FriendFeed, Tumblr, and NetFlix. While it’s running on your Apple TV, you can access the interface remotely with a browser from anywhere on your LAN at http://ipaddress:8800 assuming you have enabled the web server interface.

Hulu is another terrific resource for movies, TV shows and music videos. It is available through Boxee. There are a few ads but not many. For a lot of the movies, you’ll also need to set yourself up an account there and configure your uncrippled Apple TV accordingly.

But What About Asterisk®? We knew someone would ask. Sure. An Asterisk for Mac solution should work on the Apple TV if you don’t plan to use it as a media center. For best results, compile everything on a separate Tiger Mac, and then move it over. Keep in mind that the device is limited to 256MB of RAM so simultaneously using the Apple TV as both an Asterisk PBX and a media center more than likely will cause unacceptable performance degradation in both your phone calls and your music and video streams. Someday perhaps we’ll give it a try. In the meantime, enjoy your new open source media center!

---

Want a Bootable PBX in a Flash Drive? Next week to celebrate the beginning of Nerd Vittles’ Fifth Year, we’ll be introducing our bootable USB flash installer for PBX in a Flash with all of the goodies in the VPN in a Flash system featured a few weeks ago on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and our flash installer in less than 15 minutes!

If you’d like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment at this link with your best PBX in a Flash story.³

Be sure to include your real email address which will not be posted. The winner will be chosen by drawing an email address out of a hat (the old fashioned way!) from all of the comments posted over the next couple weeks. Good luck to everyone!

---

New Fonica Special. If you want to communicate with the rest of the telephones in the world, then you’ll need a way to route outbound calls (terminations) to their destination. For outbound calling, we recommend you establish accounts with several providers. We’ve included two of the very best! These include Joe Roper’s new service for PBX in a Flash as well as our old favorite, Vitelity. To get started with the Fonica service, just visit the web site and register. You can choose penny a minute service in the U.S. Or premium service is available for a bit more. Try both. You’ve got nothing to lose! In addition, Fonica offers some of the best international calling rates in the world. And Joe Roper has almost a decade of experience configuring and managing these services. So we have little doubt that you’ll love the service AND the support. To sign up in the USA and be charged in U.S. Dollars, sign up here. To sign up for the European Service and be charged in Euros, sign up here. See the Fonica image which tells you everything you need to know about this terrific new offering. In addition to being first rate service, Fonica is one of the least expensive and most reliable providers on the planet.
 
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.
 
 

---

Some Recent Nerd Vittles Articles of Interest…

1. Disgruntled customers reportedly have filed over a billion dollars’ worth of lawsuits over their bricked iPhones claiming Apple did it intentionally. Great PR move there, Steve! [↩]
2. The Apple TV actually runs a modified version of Tiger (aka Mac OS X 10.4). [↩]
3. This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. [↩]

About a year ago, we wrote about an incredible new whole-house audio system that is priced (literally) tens of thousands of dollars below the cost of a comparable "turnkey" system that you typically would purchase from a home audio consultant. You can read the rest of the (initial) story toward the end of today’s article. We decided to revisit the World of Sonos because there have been some incredible developments in the last 14 months, and today the company announced a new partnership with Sirius satellite radio. So here’s an update.

For those that are new to Sonos, you basically buy a little $500 Wi-Fi box for each room in your home or office where you want to play music. You plug in a pair of speakers and connect to your NAS-savvy music library. We recommend dLink’s DNS-323 which provides RAID1 mirrored SATA drives in any size you desire (about $180 delivered from NewEgg plus SATA drives). Be sure the drives you pick are on dLink’s compatibility list! If you happen to use Comcast for your broadband service, you also receive a free Rhapsody subscription which can be played (through a Windows PC) on every Sonos system in your house for free. For the rest of you, the Sonos system also supports streaming audio from more than 300 Internet radio stations, also free. And last but not least, beginning today, you can add all of the Sirius radio stations on the planet (80+ channels) to every room in your house for just $2.99 a month assuming you already have Sirius playing away in your car. If not, it’s still only $12.99 a month.

There are few companies in the world (much less the United States) that provide flawless hardware and software, free software updates (that always work), and regular updates that consistently add value to your initial purchase. Sonos is at the top of that very, very short list. Run, don’t walk, to add this system to your home or office. You’ll thank us for years to come. Enjoy!

And, our original article last year went something like this…

We thought we’d digress today and tell you about an incredible whole house audio system. Yes, there’s the iPod for private listening and there are some streaming audio solutions for those that want music in one or two rooms of a home or office. But what if you want music (different music) available in every room of your home. Well, until now, you could look at spending $20,000 to $50,000 for a very proprietary solution such as Elan’s Home Systems. It’s no accident that you won’t find any pricing on their web site.

As luck would have it, we just moved into a new home that was prewired for audio and video in eight rooms including recessed ceiling speakers in all the rooms. While this is an expensive proposition when retrofitting an older home, it’s fairly reasonable during new construction, and many builders now include it as part of the cost of a new house. The gotcha, however, is adding the multi-room amplifier, the audio devices to produce the music, and the touchpanel control units in each room. Can you spell outrageously expensive! In round numbers, you’re looking at $5,000 for installation of a suitable amplifier, $1,500 to $2,500 for each ultra-proprietary touchpanel display, and another $10,000 or more for the audio sources. These include CD jukeboxes, iPods with infrared remote access, a multi-channel XM radio receiver to the tune of $1,500 plus XM radio fees of nearly $30 a month (for three channels) forever, and loads of consulting fees at $100+ an hour. Each of the touchpanels or keypads is manually configured to match the audio components you purchase so that you can switch audio sources, adjust volume, and skip songs in each room. The double-gotcha is that despite having spent tens of thousands of dollars on this system, you have no ability to adjust anything down the road without another $100 an hour service call from the installer. So just pray they’re still around, or you’re basically stuck with your initial setup forever. A $500 magic box is used to configure the touchpanels and keypads, and, NO, you can’t buy one. It’s not sold to consumers, just dealers. Ouch!

You should be getting the picture of why we went shopping for an alternative with a bit more flexibility. That’s when we stumbled upon an incredible product called the Sonos Digital Music System. In a nutshell, you have a self-contained system unit in each room where you want music. It includes an optional amp for connection to a pair of speakers, wired and wireless networking, and a user and streaming audio interface that is as good or better than the iPod. Then you add as many touchpanel control units to select music and music sources as your budget can afford. There are also PC and Mac versions of the touchpanel which won’t cost you a dime. Each touchpanel can control every zone (aka room) in your home. What you don’t need with this system is a house prewired for audio because each unit lets you connect directly to a set of speakers or an external amplifier if desired. You also don’t need a wired network throughout your home. Only one of the Sonos units needs to be connected to a wired network. The rest of the devices automatically configure themselves to communicate wirelessly with the other system units and controllers scattered throughout your home. If you buy the starter pack with two system units including amps and one controller unit, you’re looking at $1,200 which works out to roughly $500 per system unit and about $200 for the controller. That’s roughly one tenth the cost of a functionally similar controller unit from Elan except you can configure the Sonos controller while a dealer has to configure the Elan unit … at $100 an hour.

I feel a little like the guy selling the Ginsu knives on television: "but there’s more." Boy, is there! Not only is the sound of the systems downright incredible (depending upon your speakers, of course), but the variety of available music sources is going to make you want some of these in the morning. Each system unit can stream audio from almost any music source imaginable. This includes MP3′s stored on your PC, Mac, or our latest discovery, a $150 network-attached storage (NAS) device. You also can play Shoutcast streams, either your own or those available for free over the Internet. Another option is to map a file share from a Sonos unit to a Mac or PC. It takes about 10 seconds. Sonos units also can play music from Rhapsody. And, if you’re lucky enough to be a Comcast broadband subscriber like us, a Rhapsody streaming audio subscription with about 50 music channels is yours for free! Just login to your Comcast account and download the Comcast Rhapsody software to any Windows PC. Rhapsody Stations are every bit as good as XM or Sirius channels with one important difference. There’s no additional monthly charge to Comcast customers for as many simultaneous streams as you care to play. That’s quite a contrast from Elan’s three XM streams solution which means three rooms with XM radio and no more … for $30 a month … once you buy your $1,500 Elan XM receiver. With Rhapsody, you won’t need a receiver at all, just an old clunker PC sitting in the corner with the Rhapsody application running. It can be used for other tasks as well. At the moment, we have my daughter’s game PC running Rhapsody with four simultaneous streams playing in seven zones of the house. You can double up zones with the click of a button using any Sonos controller. In addition to all these music sources, you also can connect an old-fashioned analog audio device (like a CD jukebox or an iPod) to each system unit. Music from these sources can be streamed to any combination of rooms you choose, just like traditional Shoutcast streams or Rhapsody stations. The only thing missing with analog device streams is the album art, but it still sounds great.

There are some other reviews of the Sonos system which are worth a look. Check out David Pogue’s article in the New York Times, the Home Theater View, Audioholics, Playlist Magazine, and PC Magazine. Then you’ll want to run, don’t walk, to buy at least one for yourself! You can purchase units from Sonos and most of their dealers with a 30-day money-back guarantee. We installed eight systems with four remotes in just over two hours. We haven’t quit listening since. Now you know why we’re running a little behind on the Asterisk® and TrixBox articles. Enjoy!

---

Some Recent Nerd Vittles Articles of Interest…