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The Most Versatile VoIP Provider: FREE PORTING

Introducing Incredible Pi: $35 for Asterisk, FreePBX & Google Voice Utopia

It’s Back-to-School Time in most of the United States, and the Nerd Vittles crew needed a good fall project to distract us from all of this year’s dreadful politicians. We feel like a kid in a candy store with our new $35 Raspberry Pi® Model B. Imagine squeezing a 700 mHz ARM processor with 256MB of RAM, 2 USB ports, a 10/100 Ethernet port,if an HDMI port, composite video, a separate audio jack, an SDHC card slot, and a micro USB port onto a motherboard the size of a credit card with a total weight of 1.6 ounces and a typical power draw of 3.5 watts. Thanks to Gernot and his Asterisk for Raspberry Pi project, a full LAMP stack with Asterisk® and FreePBX® gave us a head start on turning the Raspberry Pi into yet another Incredible PBX™ platform complete with Google Voice™ for free inbound and outbound calling throughout the United States and Canada.

Trust us when we say the performance of this $35 computer is nothing short of amazing. Can it do everything a $200 dual-core Atom PC can do? No. Can it do 90% of everything for someone whose requirements do not exceed a few simultaneous calls at a time but still wants a full-blown PBX for call routing, voicemail, IVRs, music on hold, and text-to-speech and speech-to-text apps for a home, a SOHO office, a Little League team, or a dorm room? Absolutely.

Call our Raspberry Pi at 1-843-284-6844 and check out the Incredible Pi Smörgåsbord for yourself.

Other Incredible Pi Resources: Quick Start Guide for Incredible Pi 3.1 and 35 Free Incredible Pi Apps Tutorial

Incredible PBX for Raspberry Pi: What’s Included

If you’ve never heard of The Incredible PBX, here’s the Top 20 for the Raspberry Pi platform. In addition to the base install with Debian 6, Asterisk 1.8, FreePBX 2.10, Apache, SendMail, MySQL, PHP, and phpMyAdmin, Incredible Pi offers the following built-in PBX components using a wired or wireless network connection:

Ordering Information: What It Takes to Get Started

A complete Incredible PBX build for the Raspberry Pi (Incredible Pi™) is finally finished, and we wanted to give you a head start on ordering the pieces you’ll need to get things going since there is a three to five week delay on some of the components due to huge demand. Nothing is ever just $35, of course. So here’s what you’ll need. The Raspberry Pi itself can be ordered from Newark or MCM. Delivery times and stock vary so check both sites. At the time we ordered, they were quoting 5 weeks. We actually got one from Newark in one week, and as of now MCM has some in stock for immediate shipping. You also need a power adapter. If you have a micro USB power brick for your cellphone, chances are it will work. As long as it’s at least 1 amp and 5 volts, you’re good to go. The recommended 700 mA adapters have reportedly caused issues for some folks so splurge and order a good one. They’re $10 from Amazon while supporting Nerd Vittles.1 If you plan to add the WiFi adapter covered below, you will need at least 1.2 amps to avoid lockups. Next, you’ll need an SDHC card which serves as your hard disk. Be very careful here. The Type 10 cards which arguably would provide the best performance turn out to be a nightmare. Many of them simply don’t work. For a basic system, we’d recommend you start with either the SanDisk 4GB or 16GB SDHC Type 4 card. The 4GB card is available almost everywhere for under $10, or you can order two from Amazon with free Prime shipping for about the same money. If you already have a USB keyboard and a monitor or TV with an available HDMI port or composite video and audio ports, then you’ve got everything you need to get started. Actually, you can do without the monitor and use SSH if you’re either using Incredible Pi 1.3 or if you can decipher the DHCP address of Incredible Pi by reviewing your firewall log. With version 1.3 and beyond, you can plug in a pair of earbuds, and the server will whisper your IP address in your ear after the boot process completes. By the way, you may also want to put your name in the queue for a Pibow case, completely unnecessary but very cute with a long waiting list. It costs almost as much as the computer.

For down the road, suffice it to say, you’re probably not going to want to run Asterisk, FreePBX, and Google Voice on a disk platform of 4 gigabytes although our demo system does exactly that. Between Debian 6.0, Asterisk 1.8, FreePBX 2.10 plus numerous Incredible PBX utilities, well over half of the available 4GB will be consumed by software. For distribution purposes, we’re providing an image that you can write directly to the SDHC card. Unfortunately, the image size determines the amount of space it will actually use on an SDHC card. We will cover below how you can install the 4GB image onto a 16GB card and then expand the size of the main partition to fill the remaining space on the card. But it wouldn’t make sense to distribute a 16GB image because of bandwidth issues. So we’d recommend you purchase a 16GB SanDisk Type 4 SDHC card for production use. Take our advice. Don’t get creative in choosing your SDHC card. Use the one that we’ve already tested and that we know works. If you want to skip the 4GB card completely, that’s perfectly fine, too. The 16GB Type 4 SanDisk card is under $10 from Amazon with free Prime 2-day shipping, or you can pay double and pick one up at your neighborhood Radio Shack store. We’ll also show you how to back up your SDHC card (any size) with your own settings and restore it onto a 16GB card without missing a beat. Unfortunately, once you migrate and expand the main partition on the 16GB card, you’re stuck with that topology. There’s no going back without starting over, and your backups will consume 16GB of storage rather than 4GB. But it sure is convenient… and easy. By the way, here’s a more reasonably priced case and here’s an awesome clear case if you just want something that’s functional.

Finally, you’re going to need an existing Linux server or Mac on which to create your bootable SDHC cards for use with Raspberry Pi. You’ll also need this server to make backups of your existing setup. DD is a wonderful low-level disk copying utility found in Linux, and we’re going to be using it a lot. PBX in a Flash with or without Incredible PBX makes an ideal platform because all of the necessary SDHC disk management utilities already are installed. If you’re not using our recommended Foxconn server, then you’ll need a server with an SDHC card slot, or you can purchase an inexpensive USB-to-SDHC adapter from Amazon for under $2. Without the case, you can build an awesome Asterisk platform for home or SOHO use for under $50, and it can handle 3 simultaneous SIP calls without noticeable degradation.

Here’s a shot of our favorite Incredible Pi setup. You’ll need the LAN cable for the initial boot so that you can SSH into the device to plug in your WiFi SSID name and password. You’ll need the earbuds for the first and second boot to decipher the IP address of the device on your LAN, wired and wireless. No monitor connection is ever required. Just log in with SSH from your Mac or PC. After the first two boots, you can dispense with the CAT5 cable and the earbuds. All of the components are covered in this article.


Creating a Bootable Incredible Pi SDHC Card

SanDisk SDHC Type 4 cards come pre-formatted so all you need to do is insert the card into the SDHC slot of your Asterisk server and reboot it. If you prefer to use a Mac, go here for the tutorial. Before you insert your SDHC card and reboot, log in as root and run the following command: fdisk -l. This will tell you what the existing disk topology of your server is. Write it down. What you don’t want to do is accidentally choose your main Linux drive as the device to copy the Incredible Pi image to, or you end up with a mess. Now insert your SDHC card and reboot your Asterisk server so your SDHC card will be visible. Then run fdisk -l again to decipher the device node of your card. It should be at the bottom of the list and will be something like /dev/sdb. We don’t use /dev/sdb1 in the dd transfer step below, just the base devnode: /dev/sdb or whatever letter your SDHC drive happens to be.

Next, using a web browser, download the latest Incredible Pi image and transfer it to the /root directory of the Linux computer you’ll be using to copy the image to your SDHC card. As we said, we recommend a PBX in a Flash server because it has all of the SDHC utilities already included. Incredible Pi is a free download from SourceForge. The feature sets are described in the SourceForge readme.txt file. Once you’ve copied the desired tarball to your Linux server, decompress it: tar zxvf incrediblepi-1.x.tgz where x is the version you downloaded. Verify the integrity of the image file using md5sum:


1.7: md5sum debian6-incrediblepi-10-09-2012.img => e601ecd890a400de2e03009c034353c2
2.1: md5sum debian7-incrediblepi-16-09-2012.img => 007efb9d2cdf86f054c5cf26c4f5de9a

Finally, run the Linux install script: ./make-sdhc. It takes about 30-60 minutes to copy the image to your SDHC card. If you have a Mac, copy make-sdhc-mac to your Downloads folder and use it.

While you’re waiting, you can read about all the latest changes and additions to Incredible Pi here.

If you’re using our recommended 16GB SanDisk Type 4 SDHC card, then what you have at this juncture is a 16GB card on which only 4GB is being used. Here’s how to expand the main partition to use the other 12GB of space on the card while logged in as root. First, we need to make sure your card’s geometry matches our card setup. From the command prompt, issue the following commands using the proper device node (/dev/sdb) of your card:

parted /dev/sdb
(parted) unit chs
(parted) print

Here’s what you should see:

If the geometry of your card doesn’t match what’s shown above, do NOT use our commands below to adjust your card. Instead, review the original tutorial explaining what needs to be done.

If your geometry matches our geometry above, issue the following commands while still in parted:

move 3 239943,0,0
[press enter to accept default End]
rm 2
mkpart primary 1232,0,0 239942,3,31
print

After entering the above commands, your new card geometry should look like this:

Now type quit. Then complete the resizing by issuing the following commands using your correct device node:

e2fsck -f /dev/sdb2
resize2fs /dev/sdb2
sync

Remove the card from your Linux machine and insert it into the SDHC card slot on the Raspberry Pi.

We’ve also found one 32GB Type 10 card that’s reliable, the Kingston Ultimate X (SD10G2/32gb) which costs about $40 at Amazon. We’ve included the resizing steps below:

fdisk -l
parted /dev/sdb
unit chs
print
move 3 483887,0,0
[press enter to accept End address]
rm 2
mkpart primary 1232,0,0 483886,3,31
print
quit
e2fsck -f /dev/sdb2
resize2fs /dev/sdb2
sync


Preparing the Raspberry Pi for Blastoff

Once the card is in place, there are a few more preliminary steps before you apply power to the device. Plug in a CAT5 LAN cable that is connected to your firewall-protected private network. Make certain that your router is handing out DHCP addresses properly since the Raspberry Pi can’t boot without a network connection to obtain the correct time from an NTP server on the Internet. It has no onboard time clock! Plug in a USB keyboard to one of the two USB ports on the Raspberry Pi. Connect either an HDMI or composite video cable between the Raspberry Pi and a monitor or TV. Finally, plug in a reliable power adapter and insert its micro USB connector into the slot on the Raspberry Pi. You should immediately see a raspberry on your screen with a scrolling list of commands that are executing while the Raspberry Pi is booting. Watch carefully. If you see a bunch of "waiting for hardware interrupt" notices, then you didn’t heed our advice on the type of SDHC card to use. Your card is not compatible for use on the Raspberry Pi so reformat it and use it with your camera. Then go buy the SanDisk card we recommended. The entire boot process should take about 30 seconds. With an incompatible card, that can stretch out to more than an hour. When the boot process completes, the IP address of your Incredible Pi should be displayed with a Linux login prompt. Write down your IP address. You’ll need it in a minute.

Securing Incredible Pi

Congratulations! You’re now ready to begin the Incredible Pi adventure. But, before we get started, first things first. Let’s secure your server. Log in with the username: pi. The password is raspberry. Now change the password to something really secure by issuing the command: passwd. Now do the same thing for the root password of the device: sudo passwd root.

Next, we need to set the default timezone on your machine for both Linux and PHP5. First, switch to the root user account so we don’t have to type sudo before each command: su root. Enter your new root password when prompted. Then run this command to set the default timezone with Debian: dpkg-reconfigure tzdata. For PHP5, you have to manually enter the timezone in the php.ini file: nano -w /etc/php5/apache2/php.ini. Press Ctrl-W to search for timezone. Be sure the line is uncommented and plug in your correct timezone. If you don’t know the magic words to use for your timezone, here’s the list. Save the file: Ctrl-X, Y, and press Enter. Now restart Apache: service apache2 restart.

First, we need to remove the default DUNDI secrets from Asterisk so fresh ones can be automatically generated when you restart Asterisk.

asterisk -rx "database del dundi secret"
asterisk -rx "database del dundi secretexpiry"
amportal restart

Now we need to do the same thing for your SSH keys:

rm /etc/ssh/ssh_host_* && dpkg-reconfigure openssh-server

Now let’s be sure Asterisk is up and running. Then you can continue your adventure using a web browser and the FreePBX GUI. Before we switch, start up the Asterisk CLI: asterisk -rvvvvvvvvvv. Verify that Flite is functioning for TTS: core show application like flite. And make sure the MySQL components are in place to support CDR reporting: module show like mysql. Finally, let’s verify that Jabber is alive and well even though we haven’t set up any Google Voice accounts yet: jabber show connections.

Finally, a few words of warning about security. Incredible Pi is designed to run behind a hardware-based firewall with no Internet exposure to the server itself. Don’t cheat! Prior to the 1.2 release, there are no security mechanisms in place: no IPtables firewall and no Fail2Ban. Beginning with Incredible Pi 1.2, the Linux firewall (iptables) is included, but our recommendation still stands unless you are using the preinstalled Travelin’ Man 3 to enable access of a remote phone connection to your PBX.

All builds do include Suhosin which has been properly configured to facilitate use of phpMyAdmin with MySQL. Within FreePBX, you can secure extensions with strong passwords and IP address filtering, and you need to do that. Exposing Incredible Pi‘s web server to Internet access would be an open invitation to an expensive phone bill. Don’t do it! You’ve been warned.

Securing Incredible Pi with Travelin’ Man 3

Travelin’ Man 3 is a collection of programs that implement firewall whitelists (safe IP addresses) using the Linux firewall, iptables. Before you ever expose your server to any kind of Internet access, read the Nerd Vittles Travelin’ Man 3 article. As delivered with Incredible Pi 1.2 and later, iptables is configured to block all access to your server except from non-routable IP addresses (typically used on LANs sitting behind hardware-based firewalls). The one exception is SIP and IAX access from VoIP Trusted Providers. This allows you to add trunks to your server from these providers without touching your firewall settings.

There are instances in which you may actually need to connect your server from a public Internet site. For example, if you travel for a living and want to use a softphone connected back to your server from a distant hotel room or customer site, you would need access through both your hardware-based firewall and iptables. If one of your children is away at school and needs a free telephone connection, this might also warrant a change in your firewalls. Because of the low cost of a Raspberry Pi, we still believe AND RECOMMEND that you use separate servers to meet remote requirements.

If you still believe remote access is necessary after reading the Travelin’ Man 3 article, then the tools are available by logging into your Incredible Pi server as root and changing to the /root directory. Here are the four apps:

For whiz kids only, Debian manages iptables quite differently than what you may be accustomed to on the CentOS platform. Debian stores iptables rules in /etc/network/iptables. You can reload the iptables rules like this: iptables-restore /etc/network/iptables. And you can display the rules currently in effect like this: iptables-save. Be careful!


Configuring Incredible Pi with FreePBX

Now we’re ready to configure Incredible Pi so that you can start making and receiving calls. We’ll be using the FreePBX web GUI. To begin, using a browser on your desktop, access Incredible Pi by pointing to the IP address of your server (that you wrote down above). Choose FreePBX Administration at the main menu. When prompted for your username and password, enter admin for both.

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

Before you do anything else, change your admin password to access FreePBX. From the main FreePBX GUI, choose Admin => Administrators. Next, set your default email address at the bottom of Settings -> General Settings.

So here’s our 7-Step Checklist to set things up. After you complete these steps, you can start making free calls throughout the U.S. and Canada. And people can call you using your new Google Voice number.

1. Create a free Google Voice account
2. Set up Extension to connect to softphone
3. Create a Google Voice Trunk using GV credentials
4. Create an Inbound Route from Google Voice to Extension
5. Create an Outbound Route from Extension to Google Voice
6. Download and Configure YATE softphone
7. Make Your First Call

Creating a Free Google Voice Account

You’ll need a dedicated Google Voice account to support Incredible Pi. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible Pi. Google Voice no longer is by invitation only so, if you’re in the U.S. or have a friend that is, head over to the Google Voice site and register. Don’t use funky characters in your Google password! If you’re living on another continent, see MisterQ’s posting for some setup tips.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it’s over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for Incredible Pi to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

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

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued with Incredible Pi.

Configuring an Extension to Use with a SIP Phone

As mentioned, an Extension is the component in FreePBX that actually interfaces with a telephone. Whether to use a hardware-based phone or a softphone is your choice. Without an adapter, POTS phones won’t work with Incredible Pi. You’ll need a SIP phone. We’ll set up a softphone today to get you started, but first we need to configure an extension in FreePBX. We’ve actually made one for you to use, extension 701. So, rather than create a new one, let’s just modify the one that’s already in place. You can add additional extensions later to support additional phones.

From the main FreePBX GUI, choose Applications -> Extensions. Then click on 701 in the Extension List on the right side of your display. You’ll see a form that looks like this:

For now, we only need to make a few changes. First, you need a very secure password for both the extension itself and your voicemail account for this extension. The extension secret needs to be a combination of letters and numbers. The Voicemail Password needs to be all numbers, preferably six or more. Replace the existing 1234secret and 1234 with your own (very secure) entries. You also need to lock down this extension so that it is only accessible from devices on your private LAN. You do that with the deny and permit entries which currently are filled with zeroes. Leave the deny entry the way it is which tells Incredible Pi to block everybody except those allowed in the permit entry below. For the permit, we need the first three octets of your private LAN address, e.g. if your LAN is 192.168.0.something then the permit entry will be 192.168.0.0/255.255.255.0.

Finally, you need to plug in your actual email address in the Voicemail section so that voicemails can be delivered to you when someone leaves a message. You can also include a pager email address if you want a text message alert with incoming voicemails. If you want the voicemails to automatically be deleted from the server after they are emailed to you (a good idea considering the disk storage limitations of an SDHC card), change the Delete Voicemail option from No to Yes. That’s it. Now save your settings by clicking the Submit button. Then reload the dialplan by clicking on the red prompt when it appears.

In case you’re curious, unless you’ve chosen to automatically delete voicemails after emailing them, you can retrieve your voicemails by dialing *98701 from any extension on your phone system. You’ll be prompted to enter the voicemail password you set up. In addition to managing your voicemails, you’ll also be given the opportunity to either return the call to the number of the person that called or to transfer the voicemail to another extension’s voicemail box. And you can always leave a voicemail for someone by dialing their extension number preceded by an asterisk, e.g. *701 would let someone leave you a voicemail without actually calling you.

Activating a Google Voice Trunk in FreePBX

To create a Trunk in FreePBX to handle calls to and from Google Voice, you’ll need three pieces of information for the Google Voice account you set up above: the 10-digit Google Voice phone number, your Google Voice account name, and your Google Voice password. Once you have these in hand, choose Other -> Google Voice from the FreePBX GUI. The following blank form will appear:

Fill in the blanks with your information and check all 3 boxes. If your Google Voice account name ends in @gmail.com, you can leave that out. Otherwise, include the full email address. Then click Submit Changes and reload your dialplan when prompted.

There’s one more step or your Google Voice account won’t work reliably with Incredible Pi! From the Linux command prompt while logged into your server as root, restart Asterisk: amportal restart

Creating an Inbound Route for Your Google Voice Trunk

Now that you’ve created your Google Voice Trunk, we need to tell FreePBX how to process the call when someone dials your Google Voice number. There are any number of choices. You could simply ring an extension. Or you could ring multiple extensions by first creating a Ring Group which is just a list of extension numbers. Or you could direct incoming calls to an Interactive Voice Response (IVR) system (we’ve actually set one up for you to play with). For the time being and since you only have one extension at the moment, let’s just route incoming Google Voice calls to extension 701.

To do this, you create an Inbound Route based upon the DID (phone number) of the Google Voice trunk. In FreePBX, choose Connectivity -> Inbound Routes. Fill in the form so that it looks like the example below using your own 10-digit Google Voice description and number instead of Atlanta GV and 6781234567. Be sure to set the Destination.

Then click Submit. But, before you reload the dialplan, make one change to the form. Click on the CID Lookup Source pull-down menu and choose CallerID Superfecta. This tells FreePBX to actually add names to phone numbers when someone calls. Now click Submit again and reload the dialplan when prompted.

Creating an Outbound Route for Google Voice Calls

FreePBX is actually smart enough to create an outbound route for your new Google Voice trunk so that you can place calls from any extension by dialing either a 10-digit number or 1 plus a 10-digit number to call anyone in the U.S. or Canada. If that’s all you care about, you can skip to the next section. But there’s more.

You can have more than one Google Voice trunk with Incredible Pi, and each one could be in a different area code. For example, you may do business in many different places and would like a local number for folks to call. Or Grandma may live in a distant city, and you’d like her to be able to call you without paying long distance charges. Then there are the kids. If you have three, you might want to give each of them their own Google Voice number which would ring just their phone. And, for outbound calls, you’d like each of them to use their own Google Voice trunk. All of these options are possible with Incredible Pi.

For outbound calls with multiple Google Voice trunks, you need a way to tell the system which trunk to use. We recommend dial prefixes that identify the city of the trunk, e.g. ATL, NYC, MIA. Or, for the kids, a dial prefix made up of initials, e.g. KHM, RWM, JSM. These dial prefixes get stripped off before the call is actually placed so the prefix is only used to determine the trunk used for placing the call.

To implement dial prefixes, you’ll need to adjust the default entries for your Google Voice Outbound Routes by adding the prefix option as an additional Dial Pattern. Here’s an example using an Atlanta Google Voice trunk where we want to allow a prefix of ATL-XXX-XXX-XXXX to force a call to go out on the Atlanta Google Voice trunk:

Another option may be appealing if you happen to make a lot of international calls and don’t want to pay for them. First, you can read all about iNum calling in this Nerd Vittles article. Another hidden feature in Google Voice is the ability to place iNum calls worldwide at no cost. To implement this, you’ll need to add another Dial Pattern to your Google Voice trunk. Prepend: 8835100 with Match Pattern: XXXXXXXX. Now you can dial iNum DIDs by dialing just the last 8 digits using any phone on your server. For example, try out the Nerd Vittles’ Dictionary Demo by calling 09901997. There also are a huge number of iNum Access Numbers that will let you call back to your server or any other iNum DID from almost anywhere in the world at no cost. These are covered in the Nerd Vittles article as well.

Configuring a YATE Softphone

As we mentioned, the easiest way to get started with Incredible Pi is to set up a YATE softphone on your Desktop computer. Versions are available at no cost for Macs, PCs, and Linux machines. Just download the appropriate one and install it from this link. Once installed, it’s a simple matter to plug in your extension 701 credentials and start making calls. Run the application and choose Settings -> Accounts and click the New button. Fill in the blanks using the IP address of Incredible Pi, 701 for your account name, and whatever password you created for the extension. Click OK.

Once you are registered to extension 701, close the Account window. Then click on YATE’s Telephony Tab and place your first call. It’s that easy!

Configuring Email Messaging

Incredible Pi comes preconfigured with SendMail to provide reliable delivery of outbound email messages. You can test it by logging in and issuing the following command using your own email address instead of yourname@gmail.com:

echo "test" | mail -s testmessage yourname@gmail.com

If you don’t receive the message, chances are that your Internet Service Provider blocks downstream mail servers from sending email to reduce spam. To fix it, the simplest way is to configure SendMail to use your ISP as the smart relay host for outbound mail, e.g. with Comcast, it’s smtp.comcast.net. It takes about 10 seconds to configure. After logging in as root, edit /etc/mail/sendmail.cf. Change line 116 from DS to DSsmtp.comcast.net using the SMTP gateway domain of your ISP. Save your change and restart SendMail: service sendmail restart. Then send another test message to make sure it works. If that fails, you can use Gmail as the mail relay. Here’s how. Now voicemail messages delivered directly to any email account by inserting your email address in the Voicemail section of the extension accepting your voicemail messages.

Configuring SMS Messaging

Incredible Pi also allows you to dictate messages and deliver them to any phone which accepts SMS messages. Just dial S-M-S (767) from any extension on your server. In addition, you can use Incredible Pi’s Message Blaster to send SMS messages to a group of individuals, e.g. a Little League team. In order to use SMS messaging, you must have at least one Google Voice account configured on your server. Then it’s a simple matter of inserting your Google Voice account name and password in two files. First, edit extensions_custom.conf in /etc/asterisk. On the second line of the 767 extension, replace acctname with your Google Voice account name (without @gmail.com). On the third line, replace acctpass with your Google Voice password for this account. Reload your dialplan to activate the new settings:

asterisk -rx "dialplan reload"

For the message blasting service, change to the /root directory. Insert the numbers to be dialed in the smslist.txt file following the example in the file. Enter the SMS message to be sent in smsmsg.txt keeping in mind that many phone providers limit SMS messages to 140 characters or less. Finally, insert your Google Voice account name (with @gmail.com) and password in smsblast. To kick off an SMS message blast, just issue the command: ./smsblast.

Configuring SAMBA for Windows Networking Support

Beginning with Incredible Pi 1.4, SAMBA is included in the distribution for transparent access using the Windows Networking Protocol from PCs, Macs, and other Linux machines. As delivered, SAMBA is deactivated. For obvious reasons, we recommend you never activate root login access to SAMBA. If you wish to enable SAMBA on your server, here are the steps while logged in as root:

  • 1. Set SAMBA password for user pi: smbpasswd -a pi
  • 2. Change Windows workgroup from WORKGROUP, if needed: nano -w /etc/samba/smb.conf
  • 3. Manually start SAMBA from command prompt: service samba start
  • 4. Set SAMBA to start on boot: rcconf and activate SAMBA option with space bar
  • 5. reboot

A Word of Warning: We’ve apparently reached the end of Memory Lane with SAMBA. Only activate it permanently (#4) after thoroughly testing it (#3) in your environment. We have found TTS IVRs in particular to be less than stable with SAMBA running.

Troubleshooting Audio Problems with Phone Calls

For most good routers/firewalls, there should be no problems connecting calls with Google Voice or SIP calls inbound or outbound. If you place a call and the audio is missing in one or both directions or your end of the call continues to ring even after the other person has answered, these are telltale signs of NAT and RTP connection issues. The quick fix is to plug in your public IP address and private LAN information under Settings -> Asterisk SIP Settings -> NAT Settings in FreePBX. If you continue to have connectivity issues, post the symptoms of your problem on the PIAF Forum and one of our helpful gurus will offer additional suggestions. Be sure to include the make and model of your router/firewall.

Using AsteriDex

Incredible Pi includes a robust phonebook application that uses MySQL for storage. You can access it with a browser by pointing to the following link using the IP address of your own server: http://192.168.0.185/asteridex4. Some entries for your favorite airlines are included to get you started. You can add, change, and delete entries under the Admin panel.

By dialing 411 from any phone on your system, you can speak the name of any entry in your AsteriDex database, and Incredible Pi will look up the name and dial the number. Try American Airlines just for fun.

Managing CallerID Superfecta

What began with our first release of CallerID Trifecta many years ago now has grown into one of the best examples of collaborative computing in the open source community. Maintained by the PBX Open Source Software Alliance, CallerID Superfecta now performs a number of functions in addition to matching names against phone numbers. Today you can display incoming call alerts and pop-ups on all sorts of devices in your home or office including XBMC, SqueezeBox, Winunciator, and many more. You can tailor CallerID Superfecta to meet your own local needs by opening the Default Superfecta tab under Other -> CallerID Superfecta in FreePBX. As with all database lookups, they take time. So keep in mind that you are trying to find the best match for inbound calls that takes the least time to retrieve corresponding CNAM information for the caller. To facilitate your search for the perfect combination, CallerID Superfecta includes a testing facility which will report the time required for each lookup. Then you can sort your lookup sources accordingly. To follow the latest developments, visit this thread on the PIAF Forums.

Adding Wireless Network Support

Particularly with a device the size of the Raspberry Pi, you may find it more convenient to place the unit on a bookshelf where a wired network connection is not feasible. This setup already is included in Incredible Pi 1.1 and later. For those using the 1.0 release, here’s how to use your Raspberry Pi wirelessly. First, order a TP-Link TL-WN722N USB 802.11n WiFi Adapter from Amazon for under $20. You won’t need a USB extender cable. This device can easily be plugged into one of the two USB slots without jeopardizing your ability to also connect a USB keyboard. Be sure your 5V power adapter is rated at 1.2 amps or greater to avoid lockups!2

We recommend the 1.3 Incredible Pi release or later if you plan to go the wireless route. All of the wireless networking components already are in place. Log into Incredible Pi as root (or su root for purists) and edit /etc/wpa.conf. Insert the SSID name and password for your wireless access point. Then reboot your server and wireless networking "just works."

Beginning with the 1.3 release, support has been added for the ultra-tiny AirLink 101 Wireless N adapter (AWLL5088). The setup process is identical to the 1.1 setup above. Just edit /etc/wpa.conf and insert the SSID name and password for your wireless access point. We strongly recommend disabling network connections that you aren’t using, e.g. eth0 and wlan0 if you go the AirLink 101 route. Just edit /etc/network/interfaces and comment out the eth0 line as well as the block of commands pertaining to wlan0. Then reboot, a process that now will be much quicker. If you plug in earbuds when you reboot, Incredible Pi 1.3 or later will read you the DHCP-assigned IP address when the boot process finishes so you no longer need a monitor. SSH can be used to connect to your server from any desktop PC or Mac.

Adding a PPTP VPN Client to Incredible Pi

If you’ve followed the Nerd Vittles tutorial and previously set up a PPTP VPN Server for your devices, then it’s pretty simple to add Incredible Pi to the mix by activating a PPTP VPN client. It’s only a few steps. You’ll need the FQDN or public IP address of your VPN server as well as a username and password for VPN access to your VPN server. Once you have those in hand, log into Incredible Pi as root.

Lest we forget to mention, you cannot log into your PPTP server from an IP address on the same private LAN so you’ll need to take your Incredible Pi device to a neighbor’s house to test this.

If you’re using Incredible Pi 1.5 or later, all of the PPTP VPN client software already is in place. Edit the connection template: nano -w /etc/ppp/peers/my-pptp-server. Insert the following text and replace myfqdn.org with the FQDN of your PPTP server, replace myname with your PPTP username, and replace mypassword with your PPTP password. Then save the file: Ctrl-X, Y, then Enter.

Now activate the PPTP VPN client. On your Incredible Pi server, run rcconf. If you’re using a release prior to 1.4, you’ll need to install rcconf first with the command: apt-get install rcconf. Scroll to the bottom of the list until you’ve highlighted pptp. Press the space bar to select it for automatic startup when you boot your server. Then tab to OK and press Enter.

To test it, issue the following command: /etc/init.d/pptp start. When you run ifconfig, you should now see a ppp0 entry:

ppp0 Link encap:Point-to-Point Protocol
UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1500 Metric:1
RX packets:0 errors:0 dropped:0 overruns:0 frame:0
TX packets:0 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:3
RX bytes:0 (0.0 B) TX bytes:0 (0.0 B)

Remember, it won’t show an IP address if the Raspberry Pi and your PPTP VPN Server are on the same subnet (like ours). Once you install your Raspberry Pi in a remote location, you now can access it at the first IP address in your reserved PPTP IP address pool.

Power Adapters: The Hidden Gotcha

We’ve learned a lot about power adapters since acquiring the Raspberry Pi. First, not all adapters are created equal. Second, the type adapter you require depends upon what you’ve plugged into those USB ports. If one of the residents is a WiFi adapter, then the power draw of the WiFi adapter can add an additional wrinkle.

Here’s what we’ve found. If you decide to use the TP-Link TL-WN722N adapter, you’ll need a power adapter rated for at least 5V, 1.2 amps. These are few and far between and many adapters rated at even higher amperages (and not U/L approved) still won’t keep your RasPi from crashing regularly. We have had good results with $9.99 RND Power Solutions 2.1A Dual USB AC Adapter so long as you use the USB port closest to the red light and leave the second USB unoccupied.

If you use the AirLink 101 WiFi adapter or no WiFi adapter, then a less expensive 5V, 1A adapter may suffice. They’re typically about half the price, but some users have reported issues. For testing results and details, see this thread.

We’ve tested these adapters, and they both work. We can also count on both hands the number of adapters we tested that fail to keep the Raspberry Pi functioning. If you have a 1.2+ amp adapter from a reputable tablet computer, that will probably work as well. These replacement adapters tend to cost $25 or more. So the choice is yours. If your Raspberry Pi experiences frequent lockups, then an underperforming power adapter is the likely culprit, not the Raspberry Pi itself.

VoIP Redundancy: Use It to Your Advantage

As much as we love Google Voice, things do go wrong from time to time. The real beauty of VoIP telephony is there’s absolutely no reason to put all of your eggs in one basket. Multiple providers don’t cost you much particularly if you don’t use them regularly. And, when things go wrong, you won’t have to scramble to continue making phone calls. Vitelity has been a generous, long-term supporter of Nerd Vittles and our projects. You’ll find a compelling offer below, and we encourage you to consider them. Here’s the Incredible PBX Top 20 Trunk List with some reasons why these providers made our short list:

  • AxVoice ($14.99/mo. Business Plan; $16.58/mo. Unlimited Calls to 45 Countries)
  • CallCentric (Good International Calling Rates; Free iNum DID)
  • DIDforSale (20 channels per DID; unlimited DID calls for $8.99/mo.)
  • ENUM
  • FlowRoute (Good International Calling Rates)
  • FreeNum
  • Future-Nine (Supports CallerID Spoofing)
  • Google Voice (Free DIDs and free U.S./Canada calling)
  • IPkall (Free SIP/IAX DIDs)
  • Les.net (Supports CallerID Spoofing; very low rates)
  • LocalPhone (Dirt-cheap DIDs and calling rates worldwide; Free iNum DID)
  • Simon Telephonics (Free SIP-to-GoogleVoice Gateway)
  • SIPgate (Free residential DIDs sometimes)
  • Skype (Free Skype-to-Skype calls worldwide)
  • Teliax (Unlimited inbound DID $5/mo.)
  • Vitelity (Our supporter and the Best in the Business!)
  • VoIPms (CallerID spoofing; Free iNum calling; Very low rates)
  • VoIPMyWay (Residential Unlimited: $15.50/mo. Business Unlimited: $40/mo.)
  • VoIPStreet (Free DID)

Making Backups of Incredible Pi

Last, but not least, you’ll need to make periodic backups of your Incredible Pi system unless you don’t mind starting over when disaster strikes. It’s easy using almost any Linux server, and it’s especially easy with a PBX in a Flash server.

To begin, shutdown your Incredible Pi server gracefully: sudo shutdown -h now. Once the display shows that the system has halted, unplug it and remove the SDHC card. Then insert the SDHC card into the slot or reader on your PIAF system and reboot the server. Log into your server as root and issue the fdisk -l command to decipher the devname of your SDHC disk, e.g. /dev/sdb. To make a backup of your SDHC card, issue the following commands using today’s date and the proper devname for your SDHC drive:

dd if=/dev/sdb of=/root/incrediblepi-08-20-2012.img
sync
gzip /root/incrediblepi-08-20-2012.img

When the process is finished, you’ll have a compressed image roughly one-third the size of your SDHC card.

Firmware and Kernel Updates

Only after making a backup, you may find it helpful to upgrade your Raspberry Pi firmware and kernel from time to time. Releases of Incredible Pi below 1.4 do not have the firmware updating tool in place. So you’ll first have to install it. Log into your server as root and issue these commands:

wget http://goo.gl/1BOfJ -O /usr/bin/rpi-update
chmod +x /usr/bin/rpi-update

You can decipher the kernel currently running on your Raspberry Pi by issuing the command: uname -a

To determine GPU’s firmware version, issue the command: /opt/vc/bin/vcgencmd version

Once the updater has been installed (and after you’ve made a backup!), you can update your Raspberry Pi’s kernel and firmware to the latest and greatest by issuing the following commands while logged in as root. The latest kernel addresses some issues with the USB ports and is worth installing: raspberrypi 3.2.27+ #66 PREEMPT Fri Aug 24 with GPU firmware version 332937.
rpi-update
reboot


Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number.

Originally published: Monday, August 20, 2012

Continue Reading… Incredible PBX for Raspberry Pi Turns 21


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! In fact, there is a thread dedicated to support of Incredible Pi. Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won’t have to wait long for an answer to your question.

Bug Fixes. A few bugs are to be expected. If you’re using Incredible Pi, please review these on the PIAF Forum. A serious problem with IPtables has now been fixed in Incredible Pi 1.5. A patch for previous versions is on the forum.


Astricon 2012. Astricon 2012 will be in Atlanta at the Sheraton beginning October 23 through October 25. We hope to see many of you there. We called Atlanta home for over 25 years so we’d love to show you around. Be sure to tug on my sleeve and mention you’d like a free PIAF Thumb Drive. We’ll have a bunch of them to pass out to our loyal supporters. Nerd Vittles readers also can save 20% on your registration by using coupon code: AC12VIT.



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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our 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…

  1. DISCLOSURE: Most Amazon referral links include a Nerd Vittles referral code so that Amazon can send us a boatload of money each month. It costs you nothing while keeping the Nerd Vittles lights burning brightly. We never recommend Amazon unless their prices or products are the best we could find at the time of publication. []
  2. The TP-Link adapter is no longer recommended due to it substantial USB power requirements. We recommend the AirLink 101 which can coexist with a 5V, 1A power supply. You must use Incredible Pi 1.3 or later. []

The Asterisk Mother Lode: Introducing Incredible PBX 4.0 with FreePBX 2.10

We thought we’d start your summer off with a bang by introducing an all-new Incredible PBX™. So what’s new? 50 Asterisk® Apps. 9-Layer Security. 20 Preconfigured VoIP Provider Trunks. 1-Click Installers for Asterisk.everything. FreePBX® 2.10. Certified Asterisk support. And what’s the same? It’s all still FREE!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Coming February 11: Incredible PBX 10 & Incredible Fax for 64-bit Asterisk 1.8 and FreePBX 2.10

We heard you. Yes, we needed Incredible PBX™ support for FreePBX 2.10 with Asterisk 1.8. And today it’s finally here. Incredible PBX 4.0 brings all of the original Asterisk applications plus more than a dozen new turnkey applications released in 2012 and an all-new level of security to protect your phone bill. The installation process is so simple a monkey could do it. You still can add Incredible Fax 2.0™ to deliver free faxing with HylaFax™ and AvantFax® in a setup process that’s as simple as pressing the Enter key. When you’re finished, you’ll have one of the open source wonders of the world with free phone calls and faxing throughout the U.S. and Canada together with almost every Asterisk application ever developed. There’s more good news. You don’t have to be smarter than a fifth grader to get any of it installed and working reliably with Asterisk. Just run the simple install script, and presto.

July Update: Incredible in the Cloud. For those that would prefer to run Incredible PBX 4.0 in the Cloud, RentPBX now is offering a pre-built image with PIAF-Brown and FreePBX 2.10 that is ready to go using servers all around the world. On your first order, Incredible in the Cloud is just $15/month using coupon code PIAF2012. Sign up at this link.

August 15 Update: Incredible Pi. We needed a back-to-school project, and this year we’ve chosen to port most of the Incredible PBX feature set to the new $35 Raspberry Pi. Thanks to the pioneering work of Gernot, this was fairly straight-forward. We’re still aiming for an early September release but, if you’d like to get a head start, you can order your device and follow our progress on the PIAF Forum.

If you’re curious why we no longer are supporting Asterisk 10, read all about it in last week’s Nerd Vittles article. Suffice it to say, if Digium is unwilling to fully support the platform, then we think it is a dead-end product. Unfortunately, this further splinters Asterisk development. In addition to long-term support (LTS) releases, "certified" (SLA) releases, and "other" releases, we now have a new category for Digium-supported modules and "community modules." To us, this signals the death knell for modules which Digium is no longer willing to actively support. Consequently, we will no longer recommend Asterisk 10 for production use. And we continue to be nervous about what the future holds for Google Voice support in Asterisk 1.8 and Asterisk 11 as well. The good news is, if you read last week’s article, you already know we have a rock-solid alternative waiting in the wings. YATE rocks! And FreeSentral is no slouch either.

The Incredible PBX 4 Inventory. For those that have never heard of The Incredible PBX, here’s the current 4.0 feature set in addition to the base install of PBX in a Flash with the CentOS 6.2, Asterisk 1.8 or Certified Asterisk 1.8, FreePBX 2.10, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Incredible Fax, Skype, NeoRouter and PPTP VPNs, and all sorts of backup solutions are still just one command away and may be installed using the scripts included with Incredible PBX 4 and PBX in a Flash. Type help-pbx and browse /root and /root/nv for dozens of one-click install scripts.

And then there’s the Incredible Freebie! As they say, "Never look a gift horse in the mouth." What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google’s free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet. The Incredible PBX Google Voice implementation provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you’ll never know you weren’t connected via Ma Bell’s overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

To get started, download the latest 32-bit PBX in a Flash 2.0.6.2 ISO from SourceForge, burn to then boot from the PIAF2 CD, choose the PIAF-Purple option to load Asterisk 1.8 or PIAF-Brown to load Certified Asterisk, and pick FreePBX 2.10 when prompted. Once the PIAF2 install is completed, just run the new Incredible PBX 4.0 installer. In less than an hour, you’ll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus over 50 terrific Asterisk applications to keep you busy exploring for months.

A Few Words About Security. Thanks to its Zero Internet Footprint™ design, Incredible PBX 4 is different. It remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no Internet port exposure to your actual server. For those needing remote telephone support, Incredible PBX loads Travelin’ Man 2 and 3 for you so your IPtables Linux Firewall can be either self-managed by end-users or set up with predefined IP addresses and FQDNs for all of your remote sites. If you’ve read about Asterisk’s latest SIP vulnerability published just last week and occurring almost as often as you tie your shoes, then you’ll understand why WhiteList-based server security has become absolutely essential. WhiteList Security means only those devices with a registered IP address in your WhiteList can get to your server’s resources. To everyone else, your server doesn’t even exist. Their only way to connect to you is with a POTS telephone and your published phone number.

For those with multiple servers to interconnect, we’ve provided one-click installers for not one but two VPN solutions: NeoRouter and PPTP. Suffice it to say, Incredible PBX has Security in Spades™: customized IPtables Linux Firewall, Fail2Ban tweaked for Asterisk security monitoring, FreePBX Extension Lockdown by IP address, randomized FreePBX extension passwords, Travelin’ Man 2 and 3 WhiteList Security, multiple VPN solutions for encrypted server-to-server communications, plus a bottom-up design focused on flawless operation behind a hardware-based firewall. You won’t find a more secure Personal Branch Exchange™ at any price.

Here’s the Incredible PBX 9-Layer Security Model:

Prerequisites. Here’s what we recommend to get started properly:

We’ve shifted gears on our recommended Atom platform for PIAF2 after excellent results with both the single-core and dual-core Atom kits manufactured by Foxconn (pictured on the left below). That’s the dLink Gaming Router on the right. Seems kinda silly to spend twice as much for a machine that you can build yourself in under 5 minutes. Basically you remove four screws, insert a Phillips screwdriver in one of the holes and gently pry the cover away from the box. Then you pop off the back by inserting a small flat-blade screwdriver, remove four more screws, slide in a solid-state drive (SSD) and a 4GB stick of notebook computer RAM, and you’re done in a couple minutes. Replace the screws and the cover, and you have a perfect PIAF2 platform with terrific performance and no moving parts for about $200. The link above will take you to the PIAF Forum thread for these machines. They go on sale almost weekly. See the right column of Nerd Vittles (just below our tweets) for this week’s special at Amazon. The dual-core Atom box typically is under $150. It could easily handle an office with 50+ employees sitting on a bookshelf with an Internet connection (wired or wireless!). No noise. Very little heat. Low power requirements. Perfect!

Installing Incredible PBX 4.0. The installation process is simple. Here are the 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple or PIAF-Brown with FreePBX 2.10 using 32-bit PIAF2 ISO
2. Run Incredible PBX 4 installer
3. Configure Google Voice and a softphone or SIP phone

Installing PBX in a Flash. Here’s a quick tutorial to get PBX in a Flash 2 installed. To use Incredible PBX 4, just install the latest 32-bit version of PBX in a Flash 2. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 6.2 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple or PIAF-Brown payload. You’ll then be prompted to choose your flavor of FreePBX. Choose FreePBX 2.10. Then set your time zone and set up a password for FreePBX access, and you’re all set. As part of the install, yum now will automatically update your operating system with the latest updates for CentOS 6.2.

You can download the 32-bit PIAF2 from SourceForge. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin. If you’ve chosen a machine without an optical drive such as the Atom boxes we recommend, then this Nerd Vittles article will show you how to make a bootable flash drive from the PIAF2 ISO.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS 6.2, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple or PIAF-Brown. In less than a minute, you’ll be prompted for the FreePBX version you wish to install. Choose FreePBX 2.10 and fill in your choices for the remaining prompts. Then have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes 30-60 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server’s IP address. You’ll need it to access FreePBX with your browser. While you’re logged in, issue the following command to make sure your IPtables firewall loads after your network is enabled:

echo "/etc/init.d/iptables restart" >> /etc/rc.d/rc.local

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box if you install it behind a hardware-based firewall as we recommend! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. If you plan to use Google Voice, you’ll need a dedicated Google Voice account to support Incredible PBX 4. If you want to use the inbound fax capabilities of Incredible Fax 2, then you’ll need an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX 4. It’s free at least through 2013. Google Voice no longer is by invitation only so, if you’re in the U.S. or have a friend that is, head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some setup tips.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it’s over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

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

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Incredible PBX 4.0 Installation. Log into your server as root and issue the following commands to run The Incredible PBX 4 installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx4.x
chmod +x incrediblepbx4.x
./incrediblepbx4.x

UPDATE: There are some new releases. Incredible PBX 10 supports 64-bit PIAF-Purple with Asterisk 1.8 and FreePBX 2.10. There also are prebuilt appliances for Amazon EC2 and VirtualBox. Incredible PBX 11 supports 32-bit PIAF-Green with Asterisk 11 and FreePBX 2.11. There also are prebuilt appliances for VirtualBox and VMware. Both Incredible PBX 10 and Incredible PBX 11 support Incredible Fax as well.

When The Incredible PBX install begins, you’ll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee. While you’re waiting just make sure that you’ve heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

FINAL STEPS. Once the Incredible PBX install completes, there are two optional steps for those that will have remote phones or users outside your firewall. Install both Travelin’ Man 2 and 3 for an ultra-secure system. Also be sure to run update-fixes before restarting your machine!

Logging in to FreePBX 2.10. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it’s maint. When prompted for the password, it’s whatever you set up as your maint password when you installed Incredible PBX 4. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here’s how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, click the Other tab and choose Google Voice. To Add a new Google Voice account, just fill out the form:

Phone number is your 10-digit Google Voice number. Username is your Google Voice account name without @gmail.com. Password is your Google Voice password. NOTE: Don’t use 2-stage password protection in this Google Voice account! Be sure to check all three boxes: Add trunk, Add routes, and Agree to TOS. Then click Submit Changes and reload FreePBX. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you’re still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you’ll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart. You’ll note that Incredible PBX 4 now automatically enables the Asterisk Gtalk and Jabber modules for you once you create a Google Voice account.

Incredible Fax 2 Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run install-incredfax2 after the Incredible PBX 4 install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. If you’d like to also add the optional OCR utility, just choose it when prompted. For complete documentation, see this Nerd Vittles article. Don’t forget that a REBOOT OF YOUR SERVER is requiredwhen the install is finished, or faxing won’t work! Then log in through the PIAF GUI using maint:password. Be sure to change your password!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the fax destination. This comes preconfigured with Incredible PBX 4. Just plug in your 10-digit Google Voice number and other entries shown in the form below. Save your entries and reload FreePBX.

Extension Password Discovery. If you’re too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we’ll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e "select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+—–+——-+
id         data
+—–+——-+
701      18016
+—–+——-+

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone such as the $50 Nortel color videophone we’ve recommended above. You’ll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you’re like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let’s try two simple tests. First, let’s place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you’ve dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here’s a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the more than 50 Asterisk applications included in your system.



Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today’s VoIP world. Start by reading our Primer on Asterisk Security. We’ve secured all of your passwords except your root password and your passwd-master password. We’re assuming you’ve put very secure passwords on those accounts as if your phone bill depended upon it. It does! There’s loads of additional documentation on the PBX in a Flash documentation web site.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust your Asterisk SIP settings. These now can be tweaked within FreePBX by choosing Settings, Asterisk SIP Settings. Just plug in your public IP address and replace 192.168.0.0 with the subnet address of your private network. Save settings and reload FreePBX.

Choosing VoIP Providers. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system… so that people can call you. Here’s how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you’re calling. If you’re in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That’s it. Congratulations! You now have a totally redundant phone system.

Incredible PBX includes preconfigured trunk setups for all of your favorite trunk providers. Just plug in your credentials and activate the trunks you need. In less than a minute, you’re done. Here’s the Incredible PBX Top 20 Trunk List with some reasons why these providers made our short list:

  • AxVoice ($14.99/mo. Business Plan; $16.58/mo. Unlimited Calls to 45 Countries)
  • CallCentric (Good International Calling Rates)
  • DIDforSale (20 channels per DID; unlimited DID calls for $8.99/mo.)
  • ENUM
  • FlowRoute (Good International Calling Rates)
  • FreeNum
  • Future-Nine (Supports CallerID Spoofing)
  • Google Voice (Free DIDs and free U.S./Canada calling)
  • IPkall (Free SIP/IAX DIDs)
  • Les.net (Supports CallerID Spoofing; very low rates)
  • LocalPhone (Dirt-cheap DIDs and calling rates worldwide; Free iNum DID)
  • Simon Telephonics (Free SIP-to-GoogleVoice Gateway)
  • SIPgate (Free residential DIDs sometimes)
  • Skype (Free Skype-to-Skype calls worldwide)
  • Teliax (Unlimited inbound DID $5/mo.)
  • Vitelity (Our supporter and the Best in the Business!)
  • VoIPms (CallerID spoofing; Free iNum calling; Very low rates)
  • VoIPMyWay (Residential Unlimited: $15.50/mo. Business Unlimited: $40/mo.)
  • VoIPStreet (Free DID)

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it’s merely a greeting. To those "in the know," it’s actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It’s quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It’s your phone bill. 😉

Configuring Email. You’re going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Settings, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Applications, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Final Word About Security. In case you couldn’t tell, security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks’ systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We’ll keep you entertained and provide immediate notification of security problems that we hear about. Finally, visit the PIAF Forums regularly. You’ll be surprised what you can learn in 10 minutes of browsing. Be safe!

Kicking the Tires. OK. That’s enough tutorial for today. Let’s play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O – Incredible PBX Demo (running on your PBX)
  • 1234*1061 – Nerd Vittles Demo via ISN FreeNum connection to NV
  • Z-I-P – Enter a five digit zip code for any U.S. weather report
  • 6-1-1 – Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 – Get the latest news and sports headlines from Yahoo News
  • T-I-D-E – Get today’s tides and lunar schedule for any U.S. port
  • F-A-X – Send a fax to an email address of your choice
  • 4-1-2 – Phonebook lookup/dialer with AsteriDex
  • M-A-I-L – Record a message and deliver it to any email address
  • C-O-N-F – Set up a MeetMe Conference on the fly
  • 1-2-3 – Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 – ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 – ODBC/AsteriDex Lookup Demo (Code: AME)
  • 3-3-3 – Look up a definition for any word or term
  • 9-4-9 – Weather forecast for any city in the world
  • 9-5-0 – Retrieve stock report by stock symbol
  • 9-5-1 – Latest Google News headlines
  • Dial *68 – Schedule a hotel-style wakeup call from any extension
  • 1-204-666-1001 – PIAF Support Conference Bridge (Conf#: 1091881)
  • 882*1061VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. We want to introduce you to the PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here’s the complete list of options:

  • A – Write the contents of the registry to registry.txt
  • B – PBX in a Flash install details
  • C – Extra programs install details
  • D – Update-fixes status and details
  • E – RPM install details
  • F – FreePBX modules install details
  • G – Incredible PBX install details
  • Q – Quit this program

And here’s a sample from an install we recently completed.


Special Thanks. It’s hard to know where to start in expressing our gratitude for all of the participants that made today’s incredibly simple-to-use product possible. To Philippe Sultan and the rest of the Asterisk development team, thank you for making Jabber jabber with Asterisk. Wish you were still involved! To Leif Madsen, thanks for your pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer, Tony Lewis, and the rest of the FreePBX development team, thanks for FreePBX 2.10 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite and all of the Google TTS and STT utilities work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications while allowing us to add dozens of new ones. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living. The new all-in-one installer is awesome. To all of our pals in the PBX Open Source Software Alliance (POSSA) that develop and maintain some our favorite Asterisk apps, you’re unbelievable! To Andrew Nagy, thanks for all you do and especially for keeping Google Voice humming along in FreePBX. And to our pal, Tom King, we couldn’t have done it without you. You rolled up your sleeves and really made CentOS 6.2 and Asterisk 1.8 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. CentOS 6 implementations of Asterisk are few and far between, and Tom has made it look incredibly easy. It wasn’t! And, last but not least, to all of our pioneers and beta testers who spent their Independence Day and many other days testing this new release, THANK YOU!

Don’t forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Monday, July 9, 2012


Support Issues. With any application as sophisticated as this one, you’re bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It’s the best Asterisk tech support site in the business, and it’s all free! In fact, there is a thread dedicated to support of Incredible PBX 4.0. Please have a look and post your support questions there. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won’t have to wait long for an answer to your question.


Weather Alert. If you’re already using the Nerd Vittles Weather Applications including Weather by ZIP Code and Weather by Airport Code, you may have noticed that the National Weather Service "improved" things over the Fourth of July holiday. Consequently, neither app worked any longer. The fixes now have been posted on the PIAF Forum and can be downloaded at your convenience. Incredible PBX 4.0 already includes the updates.


Astricon 2012. Astricon 2012 will be in Atlanta at the Sheraton beginning October 23 through October 25. We hope to see many of you there. We called Atlanta home for over 25 years so we’d love to show you around. Be sure to tug on my sleeve and mention you’d like a free PIAF Thumb Drive. We’ll have a bunch of them to pass out to our loyal supporters. Nerd Vittles readers also can save 20% on your registration by using coupon code: AC12VIT.



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…

YATE in a Flash: Rolling Your Own SIP to Google Voice Gateway for Asterisk

A few weeks ago we introduced you to Bill Simon’s SIP to Google Voice Gateway featuring YATE. This let you set up a SIP connection to your Google Voice accounts in about 5 minutes by filling out a simple web form. Today, we take it to the next plateau for those who prefer to do it yourself. With a little assistance from Bill (about 99% of the brainpower behind what you’re about to read), we’re pleased to now offer you the alternative of creating your own SIP to Google Voice Gateway. You need not share your Google Voice credentials with anybody. Meet YATE in a Flash™.

Using today’s tutorial, we’ll show you how to create a YATE in a Flash server to which you can connect as many Asterisk® servers as you like using garden-variety SIP trunks. For those that have been using one of the last half-dozen Asterisk 10 releases in which Google Voice connectivity was totally broken and for those who have languished at Asterisk 10.0.x simply to preserve Google Voice connectivity, today’s YATE alternative is a godsend because it restores the ability to make free incoming and outgoing calls in the U.S. and Canada using any flavor of Asterisk with nothing more than a SIP trunk connection to your YATE in a Flash server. We also believe it is in everyone’s best interests to pursue other Google Voice alternatives given Digium’s recent position to no longer support Gtalk and Google Voice.

If you read Malcolm Davenport’s comment in a vacuum, you’d probably come away believing that Google Voice is just too unreliable to be a supported piece of Asterisk. Funny thing is that Google Voice still works flawlessly with Asterisk 1.8, Certified Asterisk, ObiHai devices, FreeSwitch, and, of course, YATE. We’ll let you draw your own conclusions about who is responsible for the mess with Asterisk 10. Suffice it to say, if "the community" hasn’t managed to address this in 90 days, it’s probably never going to be resolved satisfactorily… and Asterisk 11 is just around the corner. So, for once, we find ourselves in total agreement with Malcolm, "building a business based on Google Voice calling using Asterisk is not something that would be recommended." YATE appears to us to be a much more satisfactory long-term solution for those that actually rely upon Google Voice.

All of the scripts today are licensed as GPL2 code, by the way, so you’re free to embellish and enhance them to meet your own needs. Please share your improvements with us so we can pass them along to "the community."

Prerequisites. Today’s design assumes you have a server running under CentOS™ 6.2. A virtual machine works fine. While YATE runs on many other operating systems, we wanted a platform that matched our existing PBX in a Flash™ and VPN in a Flash™ environment. You will also need one or more dedicated Google Voice accounts to use in conjunction with Yate in a Flash. Do NOT use a Google Voice account with a Gmail address that you already use for email, messaging, or web phone calls!

Using the original install scripts won’t work to run YATE on an existing Asterisk server. But, if you’re a true pioneer and appreciate the risks, we’ve now included scripts for BOTH dedicated server and colocated server setups so you won’t need to make any manual adjustments. Be advised that we haven’t tested colocated YATE and Asterisk under a real-world load yet to determine what impact YATE will have on the performance of an existing Asterisk server so it’s probably not a good idea to try this on your production Asterisk machine just yet. With the low cost of virtual machine environments, there’s really no reason to run YATE and Asterisk on the same machine or virtual machine. Suffice it to say, there are many issues with conflicting port assignments for telnet, sip, and iax2 as well as listening ports. While YATE is very flexible, this colocated setup still is untested.

PBX in a Flash 2.0.6.2.5 should be on the street within the next few days or weeks. With its new all-in-one design, there will be an ISO menu option allowing you to install Yate in a Flash as a standalone server with one click. Until then, we recommend using the PIAF 2.0.6.2.4 ISO and selecting the VPN in a Flash server option. This provides an ideal platform for YATE in a Flash with the added bonus of a NeoRouter VPN server and client which happens to be the perfect way to securely interconnect your PIAF and YIAF platforms via SIP.

Overview. Yate in a Flash actually consists of several scripts. For dedicated servers (meaning Asterisk is running on a separate machine), you’ll use install-yate and add-yate-user. For colocated servers (meaning Asterisk is running on the same machine), you’ll use install-yate-on-piaf and add-piaf-yate-user. As the names imply, the first script is used to actually set up your YATE in a Flash server. The second script is used to add SIP/Google Voice accounts to the YATE server. As part of the installation process, YATE is actually compiled from source code that you’ll find in /usr/src/yate on your server. Never run install-yate more than once on the same server.

To begin, you’ll need to download and untar the YIAF tarball. Then you run install-yate or install-yate-on-piaf to get YATE installed and configured. After creating and testing your Google Voice accounts at google.com/voice, you add user accounts to YATE for each existing Google Voice account you wish to activate on your YATE in a Flash server. Each time you run add-yate-user (dedicated) or add-piaf-yate-user (colocated), the script will create a new YATE user account, Google Voice account, and SIP account on your YATE server based upon your 10-digit Google Voice number. Do yourself a favor and delete the two scripts that don’t pertain to your particular setup: dedicated or colocated. Then you won’t have to worry about using the wrong ones down the road.

Once you have YATE set up and at least one account configured, then we’ll switch to your dedicated Asterisk server and use FreePBX® to add a SIP trunk, outbound route, and inbound route for each YATE account that was created. For outbound calling, we think the easiest method to take advantage of multiple Google Voice trunks is to use a different dial prefix for each account you wish to set up.

To keep it simple, in our examples today we’ll use airport codes as prefixes so we know which Google Voice trunk is actually being used to place a call, e.g. dialing ATL-404-555-1212 (285-404-555-1212) will tell FreePBX to dial out through an Atlanta Google Voice trunk and MIA-305-555-1212 (642-305-555-1212) will tell FreePBX to dial out through a Miami Google Voice trunk. Of course, the free calls can be placed to anywhere in the U.S. and Canada regardless of the Google Voice trunk you use. However, the outbound CallerID will always be the CallerID number of the Google Voice trunk being used to place the call. Before the call is actually sent via SIP to YATE for processing via Google Voice, we’ll use FreePBX to strip off the dial prefix and add a leading 1 to match the dial string format that YATE expects to see: 1NXXNXXXXXX. If you happen to be a regex genius, this could all be done on the YATE side as well, but using FreePBX makes it easy to follow:

^285\(1[0-9]\+\)$=jingle/\1@voice.google.com;line=GV40412334567;ojingle_version=0;ojingle_flags=noping;...etc.

Installing YATE. As we mentioned, until the PIAF 2.0.6.2.5 ISO is released with the option to install YATE, we recommend you download the PIAF 2.0.6.2.4 ISO and install the VPN in a Flash server from the all-in-one menu. Once you have completed the installation of VIAF, log into your server as root and issue the following commands to install YATE:

cd /root
wget http://pbxinaflash.com/YIAF.tgz
tar zxvf YIAF.tgz

If you’re installing YATE on a separate server than your Asterisk server, then issue the following command to install YATE:

/root/install-yate

If you’re installing YATE on the same server as your Asterisk server, then issue the following command to install YATE:

/root/install-yate-on-piaf

It takes about 5 minutes for YATE to compile. Once YATE is up and running, you can monitor your YATE server using telnet. If it’s running on a dedicated server, use the command: telnet 127.0.0.1 5038. If YATE is colocated on the same server as your Asterisk machine, use this command: telnet 127.0.0.1 5039. 5038 is reserved for Asterisk. Issuing the status command will tell you what’s loaded. And we’ve found it especially handy to issue the command: debug on. This lets you track everything going on with YATE without referring to the log: /var/log/yate. To exit from your telnet session, type quit. We, of course, are barely scratching the surface of what you can do with YATE. It also can be used as a full-fledged telephony engine. Here are some examples:

Just a heads up that the version of YATE being installed comes from an svn checkout several weeks ago. We zipped it up into a tarball which is downloaded as part of install-yate. With more recent builds, we have had problems with audio and the RTP stream. Until someone can sort out the issue, you’re well advised to stick with our snapshot if you want your calls to complete successfully.

Hopefully, today’s article will bring some of the YATE gurus out of the woodwork and inspire them to share their knowledge with the rest of the VoIP community. We’d be delighted to publish further articles. It’s a truly awesome platform. As I have mentioned to some of my colleagues, it reminds me of where the Asterisk community was about seven years ago. Much of the information about YATE is buried in endless threads of mailing list messages. This is an extremely difficult way to learn about and deploy a new technology. But we’re more than willing to do our part to spread the word. We’d also be happy to add a YATE Forum to the PIAF Forums so that everyone would have a searchable collection of tips in using YATE. Let us know what you think.

Configuring Google Voice. As we mentioned, you’ll need a dedicated Google Voice account for this. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now.

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively for this new SIP gateway. Head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for the SIP gateway to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

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

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued.

Next, go into Gmail for this same account and place a test call using your new Google Voice number. You’ll find the Call Phone icon in the Chat and SMS section of Gmail in the left column. Once you complete this step, be sure to log out of both Gmail and Google Voice for this account, or inbound calling will never work.

Finally, a heads up. If you are planning to use a Google Voice account that you set up previously from a different IP address, be advised that Google has some sophisticated protection mechanisms in place to deter the bad guys. As Bill Simon discovered, this may result in your not being able to connect to Google Voice from your new YIAF server. If that happens to you, follow the steps in this Google article to unlock your account.

Adding Accounts to YATE. Now that you have your Google Voice account set up and tested, we’re ready to add an account to YATE to manage it. First, be sure you have logged out of Gmail and Google Voice for the account you plan to use, or inbound calls will never make it to YATE. You’re going to need the following information to set up a new account on your YATE server:

Google Voice account name (without @gmail.com)
Google Voice account domain (usually gmail.com)
Google Voice account password
Google Voice 10-digit phone number
YATE account name will be auto-generated
YATE account password (make it very secure!)
IP address of your YATE server (unless colocated)

If you care about security, we’d strongly recommend you consider installing a NeoRouter VPN Client on both your YATE server and Asterisk server. Use the 10.0.0.x addresses for communications between the servers, and everything will be encrypted between the machines. It also greatly simplifies the firewall and security issues. If you’ve taken our advice and installed your YATE server with VPN in a Flash, then the VPN client is already in place. Just run nrclientcmd and fill in the blanks to activate it. For tips on VPN in a Flash server setup, see this article. Be sure to write down the 10.0.0.x address of your YATE server once you get the VPN client running.

To add a new account to YATE for your new Google Voice number, log into your YATE in a Flash server as root and issue the command: /root/add-yate-user (dedicated) or /root/add-piaf-yate-user (colocated). Fill in the blanks as shown above. Be sure to write down the FreePBX Trunk settings when they are displayed. You’ll need them in the next step.

Configuring FreePBX. To finish the install, you’ll need to open the FreePBX GUI on your PBX in a Flash server using a web browser. Here are the steps. If your system doesn’t already have a default inbound route pointing to Hangup, do that first: Setup -> Inbound Routes -> Add Incoming Route.

After you have the Default Inbound Route pointing to Hangup in place, only then is it advisable to Allow Anonymous SIP Calls. Any Anonymous SIP Call not handled by an Inbound Route will immediately be disconnected. You’ll find the Allow Anonymous SIP Calls option under Setup -> General Settings or Settings -> General Settings for FreePBX 2.10:

Once you have those two pieces in place, then you’re ready to Add a new SIP trunk, Outbound Route, and Inbound Route for each new Google Voice account that you add to YATE.

1. Add SIP Trunk. Choose Connectivity -> Trunks -> Add SIP Trunk and plug in the credentials that were provided when you added your Google Voice account to YATE. We recommend numbering your SIP trunks for Yate in sequential order, e.g. YIAF1, YIAF2, etc. We’re assuming YIAF1 is your Miami Google Voice trunk in this example so ignore the 843 area code. You’re smart enough to figure out your Miami Google Voice DID for yourself. This 10-digit Google Voice DID also goes on the end of the Register String after the hash tag (/) and is not shown below:

2. Add Outbound Route. Choose Connectivity -> Outbound Routes -> Add Outbound Route. Assuming this is the Outbound Route for your Miami Google Voice trunk, fill in the form in every spot we’ve placed a pink mark like this:

These dialing rules tell PBX in a Flash to dial out through the YIAF1 SIP trunk to Google Voice whenever a user dials a 10-digit or 11-digit number with the M-I-A (642) prefix. And it tells FreePBX to strip off the 642 and add a 1 (if it is missing) before sending the call to YATE. The SIP trunk settings in YIAF1 will assure that YATE places the outbound call on the Miami Google Voice trunk when it receives 1NXXNXXXXX from Asterisk.

3. Add Inbound Route. Incoming calls from the Miami Google Voice trunk will come into Asterisk as Anonymous SIP calls with the DID of the Google Voice trunk. In order to avoid an automatic Hangup, we need to create an Inbound Route for this DID. This will be the 10-digit DID of your Google Voice trunk and will match the 10-digit number on the end of the YIAF1 trunk’s Registration String. You can route these calls in any way you like on your Asterisk system, e.g. to an Extension, a Ring Group, an IVR, or whatever. Here’s an example for you to follow. Again, please ignore the non-Miami area code. We were too lazy to fix it.

So there you have it. You’re now the proud owner of your own SIP-to-GoogleVoice Gateway courtesy of YATE and Bill Simon. You can add as many Google Voice trunks as you like. And you’ll have Google Voice connectivity with Asterisk 1.8, Asterisk 10, or Certified Asterisk without ever worrying about Asterisk "improvements" that break Google Voice down the road. To add additional trunks, do the following. On the YATE side, add-yate-user. And, on the PBX in a Flash side, complete FreePBX steps 1, 2, and 3 above using the credentials provided by add-yate-user. Enjoy!

NEWS FLASH: We are pleased to announce a new YATE Forum to provide support for YATE in a Flash as well as YATE. Come visit soon!

Originally published: Monday, June 25, 2012


Trials and Tribulations of a Service Provider. We have one of the best service providers in the business. WestNic has offered exemplary service and a secure computing platform to Nerd Vittles and PBX in a Flash for many years. We consume enormous computing resources for what we pay. But the last couple weeks have been painful. First, we were on vacation when WestNic made the transition (again) to PHP 5.3. These things usually happen in the middle of the night, and this was no exception. Unfortunately, we still were running a very old, highly customized (but very secure) version of WordPress. When morning came, Nerd Vittles died. We immediately knew why because we already had experienced PHP 5.3 a few months earlier, and WestNic graciously rolled it back… just for us. Unfortunately (for us), they didn’t tell us the new drop dead date. And, yes, we should have been updating WordPress. But it’s kinda like going to the dentist. You never quite get around to it until you have to. Well, now we had to. This involved backing up and restoring Nerd Vittles to another server still running the older version of PHP. So far, so good. It took about three hours to do the three WordPress updates, but all went well. Then we moved the site back to its home, and nothing worked again. Unfortunately, this hit on a weekend, and the weekend guys claimed it was a WordPress problem. It wasn’t this time, but it took until Monday morning to get the new php.ini file sorted out to accomodate PHP 5.3. Whew!

Then came the real fun. About 25% of the threads on the PBX in a Flash Forum could not be displayed. All you got was a blank screen when you clicked on a thread. As is customary with these types of issues, the XenForo developers blamed the provider. And the provider blamed XenForo. The provider uses mod_security to protect its web sites. But the provider assured us that nothing had changed. Well, nothing in mod_security anyway. After days and days of testing and back and forth, it turned out that the provider had added a new security mechanism, suhosin, which its developer touts as the "Guardian Angel" for PHP. That may be true for providers, but not so much for folks that actually depend upon their sites working. Welcome to a new can of worms!

Having been on both sides of this fence, we can readily appreciate the dilemma of the service providers. They don’t want their servers hacked. Denying access to all users would accomplish that goal but would reduce the number of paying customers pretty dramatically. So we all try to reach that happy medium trading off a little security for a bit more access. In this case, it turned out to be a couple of suhosin settings that monitor the length of URLs. We discovered that only after running literally hundreds of tests. Since XenForo’s forum software makes extensive use of lengthy URLs to maintain compatibility with older vBulletin posts, this caused a problem. HTML requests with URLs exceeding a certain length are simply thrown in the bit bucket by suhosin. The biggest hint was sitting in the service provider’s Apache log, but we had no access to that information, and they never looked until two and a half days after we first opened a trouble ticket. No errors appeared in our logs, and users got nothing but blank pages where the subject of a post on the forum exceeded 50 characters. Fortunately, that was enough of a hint to finally resolve the problem. The unfortunate part of this story is that, without 25 years of personal IT experience plus over 100 IT gurus that visit our sites regularly, it’s doubtful this ever would have gotten resolved other than by begging the provider to turn off mod_security and suhosin for our sites, something we were unwilling to do. If something similar ever happens to you, the command you need to know is php -v. This will tell you what’s running with PHP on your host. Our provider had implied that suhosin had not yet been activated. php -v suggested just the opposite. So did their error log once they looked. The other place to start searching for configuration information is /usr/local/lib/php.ini. This will tell you how your provider has PHP configured and whether your local php.ini file is even activated. Our provider suggested more than once that our local php.ini file had been misconfigured. We’d never touched it and, in our case, the server’s php.ini file indicated that it was never activated regardless of what its contents may have contained.

We’re glad everything is fixed. We all learned more than we ever wanted to know about suhosin. Still wishing there had been a little better communications with our provider. It would have made resolution a lot easier and quicker for all concerned. It’s especially difficult to resolve thorny issues like this using service tickets with response times of half a day per message. Did we mention there is virtually no documentation on suhosin and what each of its several dozen settings actually do. Our apologies to everyone that was impacted by the service disruptions. We’re glad it’s behind us.



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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our 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…

5-Minute VoIP: Deploying a SIP to Google Voice Gateway

We’ve been big fans of Google Voice since the outset. But, with the exception of one brief week, the piece Google has always refused to put in place is a SIP gateway to make connections from VoIP devices a no-brainer. You’d think they’d do it for no other reason than economics. SIP calls are free. PSTN calls are not. Well, never mind Google. Bill Simon has done it for you, and he leveraged the same Yate toolkit that Google originally deployed. Today, we’ll show you how to spend five minutes and take advantage of the Simon Telephonics gateway to interconnect a dedicated Google Voice account with any SIP device you’d like, whether it’s an Asterisk® server, a smartphone with a free SIP client from GrooVe IP or Zoiper, a free softphone from Zoiper or X-Lite 4, or any SIP telephone. Once we’re finished today, you can use any SIP client to call your 10-digit Google Voice number through the Simon Telephonics gateway: SIP/9991234567@gvgw1.simonics.com. And you can make and receive calls throughout the U.S. and Canada using your new Google Voice number the old fashioned way, using a Plain Old Telephone. Did we mention that everything is free: the Google Voice number, the Simon Telephonics gateway connection, all of the inbound calls, and outbound calls throughout the U.S. and Canada… at least in 2012. If you take advantage of Bill’s gateway, we would encourage you to at least donate one day’s lunch money to Bill’s site to help pay the light bill.

Getting Started. The drill for today goes like this. First, you’ll create a new Google Voice account with a new phone number at google.com/voice. Next, you’ll make a test call from that number using the Gmail account associated with that same account. Then, you’ll register the Google Voice number on the Simon Telephonics gateway. Next, we’ll set up a SIP trunk on your Asterisk server for this new DID. Finally, configure any SIP client with an extension number from your Asterisk PBX, and you can start making and receiving calls using your new Google Voice number.

A Word About Security. Google doesn’t (yet) support OAuth authentication for Google Voice accounts. What this means is that you’ll have to use your actual Google Voice credentials to set up your account on the Simon Telephonics gateway. Could Bill steal your credentials? Absolutely. Will he? Absolutely not. Why? First, there’s no money in your Google Voice account so all he could do is make free calls on Google’s nickel, the same thing he could do using his own Google Voice accounts. Second, Bill is better off setting up his own accounts where you don’t share his password and the Google Voice call logs won’t tell you who he’s calling. If you’re paranoid, don’t put money in your calling account, make the account name something that could not be associated with you, and then check your call logs several times every day. Better yet, spend $50 and use an OBi110 device to set up your own private gateway where Obihai knows your credentials instead of Bill. 😉

Configuring Google Voice. As we mentioned, you’ll need a dedicated Google Voice account for this. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now.

We’ve tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively for this new SIP gateway. Head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for the SIP gateway to work its magic! Otherwise, all inbound and outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

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

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued.

Finally, go into Gmail for this same account and place a test call using your new Google Voice number. You’ll find the Call Phone icon in the Chat and SMS section of Gmail in the left column. Once you complete this step, be sure to log out of both Gmail and Google Voice for this account, or inbound calling will never work.

Registering on the Simon Telephonics Gateway. Now we’re ready to register your Google Voice account on the Simon Telephonics Gateway. Click on the link and fill in the blanks with your Google Voice account credentials and phone number. Be sure to include a 1 at the beginning of your Google Voice number! You’ll note that Google Apps email domains are supported as well as gmail.com addresses.

  • Google Voice Number19991234567
  • GV Usernamejoeschmo2468
  • GV Domaingmail.com
  • GV Passwordmightysecret
  • GV Password againmightysecret
  • Email Addressjoeschmo@yahoo.com

Check your entries carefully and then click the Add button. The only way to make changes if you screw things up is to delete the existing account by entering your original credentials to Delete the original account and then you Add a new one. So type carefully and check your work. Once your account is successfully registered, the Simon Telephonics Gateway will spit back your new SIP credentials. Write them down or take a screenshot and put them in a safe place. You’ll need them to set up your Asterisk SIP trunk. The Username will be your 11-digit Google Voice number with a GV prefix. The Secret will be a randomized string. The Registration String will be used in setting up your Asterisk SIP trunk and is in the proper format. The DID for your Inbound Route in FreePBX® will be your 11-digit Google Voice number.

  • Servergvgw1.simonics.com
  • UsernameGV19991234567
  • SecretXyzkk
  • Registration StringGV19991234567:Xyzkk@gvgw1.simonics.com/19991234567
  • Dialing FormatE.164 without + (for US calls, 11 digits starting with 1)

NOTE: Newer users may be provided an alternate gateway, e.g. gvgw2.simonics.com. You would obviously need to use whichever gateway FQDN is provided in all of the settings shown here.

Creating FreePBX SIP Trunk. Now we’re ready to create your new SIP trunk in FreePBX. Choose Add SIP Trunk and fill in the blanks as shown below with your new credentials. The Trunk Name can be any name you like. Don’t forget the 1 in Prepend for the Dialed Number Manipulation Rules! Leave the Incoming Settings blank. Be sure to add your Registration String from the credentials that were provided as part of the Simon Telephonics registration. Then Save Your Settings.

Creating FreePBX Inbound Route. Now you’ll need to add an Inbound Route to process incoming calls from the Simon Telephonics Gateway. The DID entry will be your 11-digit Google Voice number. The Destination for the incoming calls can be whatever you like: an extension, a ring group, an IVR, or any of the other available options on your server.

Creating FreePBX Outbound Route. If you want to send outbound calls out through your new Google Voice trunk, then you’ll need to add the SIP trunk to your outbound dialing rules. Just add the SIP Trunk Name you’ve defined to the Trunk Sequence for calls with the NXXNXXXXXX Dial Pattern, and you’re all set. Enjoy!

Originally published: Monday, June 11, 2012



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


whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our 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 Perfect Threesome: iNum + VoIP.ms + Google Voice

We’ve got a terrific new VoIP development for you today especially for those who travel internationally. For several years, a VoIP company called VoxBone has been pushing hard to establish an International Number™ (iNum™) for every phone on the planet so that every telephone could call every other telephone at little or no cost. They’re not quite there, but two recent events will certainly hasten the implementation. The first was an announcement from VoIP.ms that they would provide a free iNum DID and free iNum calling to every one of their customers with a credit balance in their account. The second was last week’s announcement from Google that they, too, would support free iNum calling worldwide using any Google Voice account. Today, we’ll show you how to take advantage of these two developments to begin making free calls worldwide using your PBX in a Flash™ server, a WiFi-enabled smartphone, and an available WiFi connection. Basically, the plan is to use free iNum calling to get back to your PBX for dial tone and then use DISA for free Google Voice calling in the U.S. and Canada.

Until everyone has an iNum or Google opens up Google Voice outside North America, the hidden beauty of iNum for those of us who have both is the cost savings that can be achieved by phoning home with iNum from anywhere in the world for free. And, once the call hits your Asterisk® PBX, it’s incredibly simple to route the call to DISA, prompt for a password, and then place a call to anywhere in the U.S. or Canada at no cost with PIAF2™ and Google Voice.

This can be accomplished in several ways. First, you can download a SIP phone and use it in conjunction with your VoIP.ms account and a smartphone to make free iNum calls from any WiFi hotspot in the world. Bria is our favorite on both the iPhone/iPad and Android platforms. If $10 is too rich for your blood, there are some free alternatives: CSipSimple for Android and 3CXPhone for Android or iPhone. A second alternative is to use Google Voice or Gtalk to connect back to your PIAF2 server via iNum and then use DISA and your local trunks to place outbound calls. A final alternative is to take advantage of the numerous local numbers now available in many countries to phone home using iNum. The only cost of these calls is the cost associated with calling the local number. You’ll find a list of the local phone numbers to make these calls on the iNum web site or in the footnote to this article.1 So today we’ll show you how to set up your PIAF2 server to support free iNum calling. It’s a 15-minute project.

VoIP.ms Setup. To get started, if you’re not already a customer, register for a voip.ms account by filling out their registration form.

Once you submit the form, you’ll have to confirm your registration by clicking on the link that is emailed to you. Then you’re ready to login with your email address and the password you set up when you created your account. That’ll bring you to the Main Portal Page for your new voip.ms account.

You’ll need a positive balance in your VoIP.ms account in order to create your free iNum account so deposit some money using PayPal or a credit card by clicking Finances, Add Funds. The minimum deposit is $25 which can be used to make penny a minute calls in the U.S. and Canada or equally reasonable calls to any phone number in the world. We won’t be doing any of that today. For today, all of our calls will be free thanks to iNum and the generous support of VoIP.ms. But the nest egg will be there as a backup to your other PIAF2 VoIP providers which is an excellent idea anyway.

Like Vitelity, VoIP.ms lets you create subaccounts to compartmentalize your VoIP services. This makes it easy to use VoIP.ms on multiple PIAF2 servers or even standalone SIP telephones. It also provides added security by separating out account names and passwords for VoIP services from your main VoIP.ms portal account that let’s you manage your settings and VoIP funding, a very good idea. So let’s first set up an account to use with Asterisk just to show you how easy it is.

From the Main Portal Menu, click on Subaccounts, Create Subaccount. The Subaccount creation form will display. Fill it out so it looks something like this. Just click on the form below to enlarge it if you want a better view.

Once you’ve clicked the button to create the subaccount, it takes about a minute for voip.ms to activate it. Then click Main Menu, Portal Home. The bottom of the portal page will now show your subaccount.

Let’s create one more subaccount. We’ll use this one so that we can access VoIP.ms from a standard SIP app running on our iPhone or Android device. We can use the subaccount either to make outbound calls directly from VoIP.ms on a pay per minute basis, or we can use it to make free iNum calls. To create the subaccount, repeat the process above and fill in the blanks using your own credentials and a very secure password. Be sure to choose ATA device, IP Phone or Softphone for the Device Type. We always leave International Calls Disabled unless we really plan to make international calls. This will not affect your ability to make iNum calls, and it reduces your financial exposure in the event your subaccount is compromised. Never, ever use auto-replenishment from your credit card on a VoIP provider account from any provider.

Before we get too far along, let’s activate your new iNum DID. Click on DID Numbers, Order DID. When the DID Order Form displays, click on the iNum link to order your free iNum DID.

When the iNum DID order form displays, fill out the form by clicking on the POP location nearest to your server. Then, in the SIP/IAX Routing column, be sure to select the Subaccount we created previously rather than the default Main Account. Finally click the Click Here to Order button.

You’ll get a Confirmation display that shows your new iNum DID. Write it down! We’ve already set up the proper routing for your new iNum DID in the previous step so you can ignore the Managing Your DID message.

That completes the setup of your VoIP.ms account with your free iNum DID. Now let’s configure your PBX in a Flash server to support VoIP.ms and iNum. We’re assuming you already have a PBX in a Flash server configured with at least one Google Voice account activated. If not, stop here and complete that step using the PIAF2 tutorial and optionally the Incredible PBX 3 and Incredible Fax 2 tutorial.

Smartphone SIP Client Setup. We used the free cSipSimple Android app to set up a connection with our second subaccount at VoIP.ms using cSipSimple’s Basic Setup Wizard. Here are the entries required to gain connectivity:

Once your SIP client is connected to VoIP.ms through your smartphone, you can make free iNum calls using this dial syntax: 0118835100xxxxxxxx where xxxxxxxx is the last 8 digits of your iNum beginning with 0. As noted previously, you do NOT have to enable international calls on your VoIP.ms subaccount for these calls to go through.

PBX in a Flash iNum Setup. We’ll be using the FreePBX GUI to configure PBX in a Flash to support iNum. Using your browser, log into the IP address of your server: http://ipaddress/admin. When prompted for your username and password, use maint and whatever FreePBX password you assigned when your server was set up.

To simplify things, we’re going to set up 2 trunks: one for your VoIP.ms subaccount and another for iNum. Begin by choosing Trunks, Add SIP Trunk in the FreePBX GUI. For Trunk Name, use voipms. For Maximum Channels, choose 2. For the Dial Pattern, enter 1 | NXXNXXXXXX and, in Outgoing Settings for the PEER Details, enter the following using your subaccount name and password as well as the POP you chose for your subaccount:

canreinvite=yes
nat=yes
context=from-trunk
host=atlanta.voip.ms
secret=subacctpw
type=peer
username=137786_myinum
disallow=all
allow=ulaw
fromuser=137786_myinum
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

Leave all the fields for Incoming Settings blank. For the Registration String, the syntax is subacctname:subacctpw@atlanta.voip.ms:5060/8835100xxxxxxxx. Using our example and assuming you’re using the Atlanta POP, the entry would look like this where xxxxxxxx is your own 8-digit iNum beginning with 0:

137786_myinum:secretPassword21@atlanta.voip.ms:5060/8835100xxxxxxxx

Verify that your server got a successful registration with your VoIP.ms subaccount by clicking Tools, Asterisk Info, SIP Info.

Now click Setup, Trunks, Add Custom Trunk. For Trunk Name, use iNum. For Maximum Channels, choose 5. For Dial Pattern, use 0XXXXXX. including the period! For Custom Dial String, use SIP/0118835100$OUTNUM$@voipms.

Next, we need to create an Inbound Route. Use your full iNum DID number in the DID Number field, e.g. 8835100xxxxxxxx where xxxxxxxx is your personal iNum beginning with a 0. Activate CallerID Superfecta for the CID Lookup Source. And choose a Destination for the incoming iNum calls. This could be an extension, an IVR, or whatever else you’ve set up on your server. For now, route it to a working extension on your PBX so we can test it below. Then you can edit the inbound route and change it to any destination.

Finally, create an Outbound Route. Name the route OutiNum. For the Dial Pattern, use 0XXXXXX. with the trailing period. For the Trunk Sequence for Matched Routes, choose inum. After you save the trunk settings, move it to the top of your trunk listing in the right column of FreePBX. What this route does is allow you to call other iNum numbers (including your own) by simply dialing the last 8-digits of any iNum that begins with 8835100 or 0118835100. These 8 digits will ALWAYS begin with a 0.

Now let’s modify at least one of your existing Google Voice Outbound Routes so that you also can make iNUM calls with Google Voice by dialing from any extension using the full 8835100xxxxxxxx international number. Go to Outbound Routes and click on the name of one of your Google Voice trunks. Add the following new Dial Pattern and click Submit Changes: 8835100XXXXXXXX

Taking iNum for a Spin. To test things out, use a phone connected to an extension other than the one you chose to route incoming iNum calls to above. Dial the last 8 digits of your own iNum DID, and that extension should begin ringing. Answer the other extension and make sure you have audio in both directions. Next, dial your complete iNum DID beginning with 8835100. This should also cause the other extension to ring even though the call was initiated through your Google Voice trunk. If you’d like to get a Weather Report by Zip Code, we’ve set up an iNum for you to try. Just dial 09901997.
Enjoy!

Originally published: Monday, February 27, 2012



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.


 

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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.

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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…

  1. Local iNum Access Numbers include the following: []

Introducing Incredible PBX 3.0 and Incredible Fax 2.0

As Nerd Vittles begins its seventh year, a birthday bash is certainly in order. And today we have not one but two of Tom King's reworked masterpieces to introduce. The PIAF2™ introduction with CentOS 6.2™ and your choice of Asterisk® and FreePBX® versions has certainly brought its share of challenges. But, with the new year, we're finally comfortable recommending everyone make the switch. Almost everything is faster, more stable, and smoother with CentOS 6.2. Yes, the pain is worth the gain. But this new platform also meant significant rewrites of some of our VoIP workhorses, and today everything is finally ready for prime time.

News Flash: Incredible PBX 4.0 is now available with FreePBX 2.10 support!

Coming January 19: Incredible PBX 11 & Incredible Fax for Asterisk 11 and FreePBX 2.11

Incredible PBX 3.0™ brings literally dozens of turnkey Asterisk applications to your PIAF2 server, and the installation process is so simple a monkey could do it. And Incredible Fax 2.0™ delivers free faxing with HylaFax™ and AvantFax® in a setup process that's as simple as pressing the Enter key. When you're finished, you'll have one of the open source wonders of the world with free phone calls and faxing throughout the U.S. and Canada together with almost every Asterisk application ever developed. There's more good news. You don't have to be smarter than a fifth grader to get any of it installed and working reliably with Asterisk. In fact, all of the new installers now are rolled into the base PBX in a Flash 2.0™ installation. Just run two simple scripts, and presto. You're done!

The Incredible PBX 3 Inventory. For those that have never heard of The Incredible PBX, here's the current 3.0 feature set in addition to the base install of PBX in a Flash with the CentOS 6.2, Asterisk 1.8 or 10, FreePBX 2.9, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Incredible Fax, Hamachi VPN, and Mondo Backups are still just one command away and may be installed using the scripts included with Incredible PBX 3.

What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

Just download the latest 32-bit or 64-bit PBX in a Flash 2.0.6.2 ISO from SourceForge, burn to then boot from the PIAF2 CD, choose the PIAF-Purple option to load Asterisk 1.8 or PIAF-Red to load Asterisk 10, and pick FreePBX 2.9 when prompted. Once the PIAF2 install is completed, just run the new Incredible PBX 3.0 installer: install-incredpbx3. In less than an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep you busy exploring for months.

Thanks to its Zero Internet Footprint™ design, Incredible PBX 3 remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

Did we mention that all of this telephone goodness is still absolutely FREE!

Prerequisites. Here's what we recommend to get started properly:

Installing Incredible PBX 3.0. The installation process is simple and straight-forward. We're down to 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple & FreePBX 2.9 using the PIAF2 ISO
2. Run Incredible PBX 3 installer
3. Configure Google Voice and a softphone or SIP phone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash 2 installed. To use Incredible PBX 3, just install the latest 32-bit or 64-bit version of PBX in a Flash 2. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 6.2 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple or PIAF-Red payload. You'll then be prompted to choose your flavor of FreePBX. Choose FreePBX 2.9. Then set your time zone and set up a password for FreePBX access, and you're all set. As part of the install, yum now will automatically update your operating system with the latest updates for CentOS 6.2.

You can download the 32-bit PIAF2 from SourceForge. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS 6.2, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple. In less than a minute, you'll be prompted for the FreePBX version you wish to install. Choose FreePBX 2.9 and fill in your choices for the remaining prompts. Then have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes 30-60 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server's IP address. You'll need it to access FreePBX with your browser.

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. You'll need a dedicated Google Voice account to support Incredible PBX 3. If you plan to use the inbound fax capabilities of Incredible Fax 2, then you'll want an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

We've tested this extensively using an existing Gmail account, and inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So, be reasonable. Do it our way! Set up a dedicated Gmail and Google Voice account, and use it exclusively with Incredible PBX 3. Google Voice no longer is by invitation only so, if you're in the U.S. or have a friend that is, head over to the Google Voice site and register. If you're living on another continent, see MisterQ's posting for some tips on getting set up.

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work... in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That's the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don't see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

  • Call Screening - OFF
  • Call Presentation - OFF
  • Caller ID (In) - Display Caller's Number
  • Caller ID (Out) - Don't Change Anything
  • Do Not Disturb - OFF
  • Call Options (Enable Recording) - OFF
  • Global Spam Filtering - ON

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Incredible PBX 3.0 Installation. Log into your server as root and issue the following commands to run The Incredible PBX 3 installer:

install-incredpbx3

When The Incredible PBX install begins, you'll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee. While you're waiting just make sure that you've heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

FINAL STEP. Once the Incredible PBX install completes, be sure to download the latest updates and patches for PBX in a Flash and Incredible. Just issue the following commands:

update-programs
update-fixes

Logging in to FreePBX 2.9. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX 3. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here's how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, click the Setup tab and choose Google Voice in the Third Party Addons. To Add a new Google Voice account, just fill out the form:

Phone number is your 10-digit Google Voice number. Username is your Google Voice account name without @gmail.com. NOTE: You must use a Gmail.com address in the current version of this module! Password is your Google Voice password. NOTE: Don't use 2-stage password protection in this Google Voice account! Be sure to check all three boxes: Add trunk, Add routes, and Agree to TOS. Then click Submit Changes and reload FreePBX. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you're still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you'll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

Incredible Fax 2 Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run install-incredfax2 after the Incredible PBX 3 install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. If you'd like to also add the optional OCR utility, just choose it when prompted. For complete documentation, see this Nerd Vittles article. Don't forget that a REBOOT OF YOUR SERVER is requiredwhen the install is finished, or faxing won't work! Then log in through the PIAF GUI using maint:password. Be sure to change your password!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the FAX Misc. Destination that comes preconfigured with Incredible PBX 3. Substitute your 10-digit Google Voice number for the DID number shown below. Save your entries and reload FreePBX.

Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone such as the $50 Nortel color videophone we've recommended above. You'll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you're like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password. We're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.

We've also included Trunk configurations for a dozen of our favorite hosting providers to get you started. You can sign up for service with any of them, insert your credentials in the existing trunk, uncheck the Disable Trunk checkbox, and then adjust your outbound route and add an inbound route for your new DID (if you get one).

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉

Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!

Enabling Google Voicemail. Some have requested a way to retain Google's voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you'll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart

;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)

But I Don't Want to Use Google Voice. If you'd prefer not to use Google Voice at all with PBX in a Flash, that's okay, too. Here's how to disable it and avoid the chatter in the Asterisk CLI. Log into your server as root and edit /etc/asterisk/modules.conf. Change the first three lines in the [modules] context so that they look like this. Then restart Asterisk: amportal restart.

autoload=yes
noload => res_jabber.so
noload => chan_gtalk.so

There's now a patch that automatically adjusts Asterisk to accommodate Google Voice whenever you have added Google Voice extensions to your system. To download and install the patch, visit the PIAF Forum.

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PIAF Support Conference Bridge (Conf#: 1061)
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:

  • A - Write the contents of the registry to registry.txt
  • B - PBX in a Flash install details
  • C - Extra programs install details
  • D - Update-fixes status and details
  • E - RPM install details
  • F - FreePBX modules install details
  • G - Incredible PBX install details
  • Q - Quit this program

And here's a sample from an install we recently completed.


Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. To Philippe Sultan and the rest of the Asterisk development team, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer, Tony Lewis, and the rest of the FreePBX development team, thanks for FreePBX 2.9 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 6 and Asterisk 1.8 and 10 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. You won't find another CentOS 6 implementation of Asterisk, and Tom has made it look incredibly easy. It wasn't! And, last but not least, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

Additional Goodies. Be sure to log into your server as root and look through all of the free scripts that are included. Just type: help-pbx.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Monday, January 23, 2012


Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.



Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.


 

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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.
 


  1. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB CD/DVD drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. You also can run Incredible PBX 3 on a virtual machine such as the free Proxmox server. A security vulnerability has been reported in the Proxmox browser so be sure to run your server behind a secure, hardware-based firewall with no port exposure to the actual Proxmox server from the Internet. []

Introducing PIAF2 and Incredible PBX 2.9 with CentOS 6.1


We're pleased to introduce the latest and greatest PBX in a Flash™ 2.0.6.1 featuring CentOS® 6.11 and the brand-new Incredible PBX™ 2.9 with an incomparable VoIP feature set. PIAF2™ provides turnkey installs of Asterisk® 1.8 or 2.0 with your choice of FreePBX® 2.8, 2.9, or 2.10. And, for those choosing to install Incredible PBX 2.9, it's been engineered to work flawlessly with the 32-bit version of PIAF2 using Asterisk 1.8 and FreePBX 2.9. For the ultimate in performance, a 64-bit version of PIAF2 is also available; however, because of its size, a DVD is required to burn the ISO. And, as noted, it is not compatible with Incredible PBX 2.9.

12/17 Update: Shortly after PIAF2 hit the street, Digium released Asterisk 1.8.8.0 and the first non-beta version of Asterisk 10. New 32-bit and 64-bit PIAF 2.0.6.1.2 ISOs will be available on SourceForge today that incorporate these new builds. In addition, a CentOS 6 video quirk has been identified on some Atom hardware. So the new ISOs include an install option to disable the problematic video testing by kicking off the install with one of the following commands instead of merely pressing the Enter key: ks-nomode, ksraid-nomode, or kslvm-nomode. You'll know if you have the problem if your server locks up. 😉 Finally, because there now are multiple stable versions of Asterisk, we have added the option to selectively choose a version of Asterisk to install. Instead of picking PIAF-Purple or PIAF-Red, you can drop down to the Linux command prompt, log in as root, and issue a command using the following syntax: piafdl -p beta_1872_purple.

Photo courtesy of mashable.com

Free Google Voice calling in the U.S. and Canada has been extended for calendar year 2012 and now can be configured using the simple FreePBX 2.9 GUI. And you can use it with or without Incredible PBX. Set up one or many Google Voice connections in less than 10 seconds per line. With Incredible PBX, we've also included Andrew Nagy's terrific EndPoint Manager that lets you configure dozens of SIP phones with the click of a button. You'll also find Kennonsoft's terrific new PBX in a Flash UI with HTML5 and CSS3 support for the latest Firefox, Chrome, and IE browsers. And, of course, you still get almost every Asterisk application on the planet preconfigured and ready to use.

With PIAF2, the installation process has been streamlined considerably. At the outset, you will be prompted for some basic information and a root password. Once the CentOS 6.1 install completes and you remove the CD/DVD during the server reboot, you will be prompted for whether you wish to tailor Asterisk using menuconfig, your time zone, the version of FreePBX you wish to install, and your master password for FreePBX access. Once you've answered these few questions, you can kick off the PIAF2 install and walk away. Depending upon the performance of your server, come back in 15-30 minutes. While it's not the quickest install on the planet, it will always be the most current because PIAF2 always loads the latest patches to CentOS as well as Asterisk and FreePBX. In other words, it's worth the wait to know you're installing a secure and up-to-date system. And, as your high school girlfriend probably taught you, faster is not always better.

The Incredible PBX 2.9 Inventory. For those that have never heard of The Incredible PBX, here's the current 2.9 feature set in addition to the base install of PBX in a Flash with the CentOS 6.1, Asterisk 1.8, FreePBX 2.9, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Faxing, Hamachi VPN, and Mondo Backups are still just one command away and may be installed using the scripts included with base Incredible PBX 2.9 installation.

Update: Incredible Fax is not yet compatible with PIAF2, but we're working on it.

What began as a kludgey, dual-call, dual-provider Google Voice implementation to take advantage of Google's free PSTN calling in the U.S. and Canada with Asterisk 1.4 and 1.6 is now a zippy-quick, Gtalk-based calling platform that rivals the best SIP-to-SIP calls on the planet and provides virtually instantaneous PSTN connections to almost anybody, anywhere. Trust us! Except for the price which is still free, you'll never know you weren't connected via Ma Bell's overpriced long-distance lines and neither will the Little Mrs. And, yes, our recommended $50 Nortel SIP videophone is plug-and-play.

Just download the latest 32-bit PBX in a Flash 2.0.6.1 ISO from SourceForge, burn to then boot from the PIAF2 CD, choose the PIAF-Purple option to load Asterisk 1.8, and pick FreePBX 2.9 when prompted. Once the PIAF2 install is completed, just run the new Incredible PBX 2.9 installer. In less than an hour, you'll have a turnkey PBX with a local phone number and free calling in the U.S. and Canada via your own Google Voice account plus dozens and dozens of terrific Asterisk applications to keep you busy exploring for months.

Thanks to its Zero Internet Footprint™ design, Incredible PBX 2.9 remains the most secure Asterisk-based PBX around. What this means is The Incredible PBX™ has been engineered to sit safely behind a NAT-based, hardware firewall with no port exposure to your actual server. And you won't find a more full-featured Personal Branch Exchange™ at any price.

Did we mention that all of this telephone goodness is still absolutely FREE!

Prerequisites. Here's what we recommend to get started properly:

Installing Incredible PBX 2.9. The installation process is simple and straight-forward. We're down to 3 Easy Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls immediately:

1. Install PIAF-Purple & FreePBX 2.9 using 32-bit PIAF2 ISO
2. Download & run Incredible PBX 2.9 installer
3. Configure Google Voice and a softphone or SIP telephone

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash 2.0 installed. To use Incredible PBX 2.9, just install the latest 32-bit version of PBX in a Flash 2.0. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 6.1 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the PIAF-Purple payload to get the latest Asterisk 1.8. You'll then be prompted to choose your flavor of FreePBX. Choose FreePBX 2.9. Then set your time zone and set up a password for FreePBX access, and you're all set. As part of the install, yum now will automatically update your operating system to CentOS 6.2 minus the 6.2 kernel.

You can download the 32-bit PIAF2 from SourceForge. Burn the ISO to a CD. Then boot from the installation CD and press the Enter key to begin.

WARNING: This install will completely erase, repartition, and reformat EVERY DISK (including USB flash drives) connected to your system so disable any disk you wish to preserve AND remove any USB flash drives! Press Ctrl-C to cancel.

At the keyboard prompt, tab to OK and press Enter. At the time zone prompt, tab once, highlight your time zone, tab to OK and press Enter. At the password prompt, make up a VERY secure root password. Type it twice. Tab to OK, press Enter. Get a cup of coffee. Come back in about 5 minutes. When the system has installed CentOS 6.1, it will reboot. Remove the CD promptly. After the reboot, choose PIAF-Purple. In less than a minute, you'll be prompted for the FreePBX version you wish to install. Choose 2.9 and fill in your choices for the remaining prompts. Then have a 15-minute cup of coffee. After installation is complete, the machine will reboot a second time. You now have a PBX in a Flash base install. On a stand-alone machine, it takes 30-60 minutes. On a virtual machine, it takes about half that time. Log into your server with your root password and write down the server's IP address. You'll need it to access FreePBX with your browser.

NOTE: For previous users of PBX in a Flash, be aware that this new version automatically runs update-programs, update-fixes, and passwd-master for you. So your system is relatively secure out of the box! See the Proxmox cautionary alert in the footnotes to this article!

Configuring Google Voice. You'll need a dedicated Google Voice account to support Incredible PBX 2.9. If you plan to use the inbound fax capabilities of Incredible PBX 2.9, then you'll want an additional Google Voice line that can be routed to the FAX miscellaneous destination using FreePBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!

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

You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work... in either direction. Google used to permit outbound Gtalk calls using a fake CallerID, but that obviously led to abuse so it's over! You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don't skip this step either. Just enter the provided 2-digit confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you'd like in Settings, Voice Setting, Phones. But...

IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That's the destination we need for The Incredible PBX to work its magic! Otherwise, all inbound and outbound calls will fail. If you don't see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings.

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

  • Call Screening - OFF
  • Call Presentation - OFF
  • Caller ID (In) - Display Caller's Number
  • Caller ID (Out) - Don't Change Anything
  • Do Not Disturb - OFF
  • Call Options (Enable Recording) - OFF
  • Global Spam Filtering - ON

Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.

Incredible PBX 2.9 Installation. Log into your server as root and issue the following commands to download and run The Incredible PBX installer:

cd /root
wget http://incrediblepbx.com/incrediblepbx29.x
chmod +x incrediblepbx29.x
./incrediblepbx29.x

When The Incredible PBX install begins, you'll be prompted for your FreePBX maint password. This is required to properly configure CallerID Superfecta for you. Your credentials never leave your server!

Now have another 15-minute cup of coffee, and consider a modest donation to Nerd Vittles... for all of our hard work. 😉 You'll find a link at the top of the page. While you're waiting just make sure that you've heeded our advice and installed your server behind a hardware-based firewall. No ports need to be opened on your firewall to support Incredible PBX. Leave it that way!

One final word of caution is in order regardless of your choice of providers: Do NOT use special characters in any provider passwords, or nothing will work!

Logging in to FreePBX 2.9. Using a web browser, you access the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Click on the Users tab. It will change to Admin. Now click the FreePBX button. When prompted for a username, it's maint. When prompted for the password, it's whatever you set up as your maint password when you installed Incredible PBX 2.9. If you forget it, you can always reset it by logging into your server as root and running passwd-master.

Configuring Google Voice Trunks in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize your Incredible PBX to meet your specific needs. If you plan to use Google Voice, here's how to quickly configure one or more Google Voice trunks within FreePBX. After logging into FreePBX with your browser, click the Setup tab and choose Google Voice in the Third Party Addons. To Add a new Google Voice account, just fill out the form:

Phone number is your 10-digit Google Voice number. Username is your Google Voice account name without @gmail.com. NOTE: You must use a Gmail.com address in the current version of this module! Password is your Google Voice password. NOTE: Don't use 2-stage password protection in this Google Voice account! Be sure to check all three boxes: Add trunk, Add routes, and Agree to TOS. Then click Submit Changes and reload FreePBX. You can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill.

While you're still in FreePBX, choose Setup, Extensions, and click on the 701 extension. Write down your extension password which you'll need to configure a phone in a minute.

IMPORTANT LAST STEP: Google Voice will not work unless you restart Asterisk from the Linux command line at this juncture. Using SSH, log into your server as root and issue the following command: amportal restart.

Incredible Fax Installation. If you want the added convenience of having your Incredible PBX double as a free fax machine, run /root/incrediblefax.sh shell script when the Incredible PBX install completes. Plug in your email address for delivery of incoming faxes and enter your home area code when prompted. For every other prompt, just press the Enter key. For complete documentation, see this Nerd Vittles article. Don't forget to REBOOT YOUR SERVER when the install is finished, or faxing won't work!

Also be sure to set up a second, dedicated Google Voice number if you want support for inbound faxing. Once the Google Voice credentials are configured in FreePBX for the additional Google Voice line, simply add an Inbound Route for this DID to point to the FAX misc. destination that comes preconfigured with Incredible PBX 2.9. Just substitute your 10-digit Google Voice number for the DID number shown below. Save your entries and reload FreePBX.

Extension Password Discovery. If you're too lazy to look up your extension 701 password using the FreePBX GUI, you can log into your server as root and issue the following command to obtain the password for extension 701 which we'll need to configure your softphone or color videophone in the next step:

mysql -uroot -ppassw0rd -e"select id,data from asterisk.sip where id='701' and keyword='secret'"

The result will look something like the following where 701 is the extension and 18016 is the randomly-generated extension password exclusively for your Incredible PBX:

+-----+-------+
id         data
+-----+-------+
701      18016
+-----+-------+

Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you'll want a real SIP telephone such as the $50 Nortel color videophone we've recommended above. You'll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you're like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let's download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using your actual password for extension 701 and the actual IP address of your Incredible PBX server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations!

Here's a brief video demonstration showing how to set up a softphone to use with your Incredible PBX, and it also walks you through several of the dozens of Asterisk applications included in your system.

Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk with the command: amportal restart.

Learn First. Explore Second. Even though the installation process has been completed, we strongly recommend you do some reading before you begin your VoIP adventure. VoIP PBX systems have become a favorite target of the hackers and crackers around the world and, unless you have an unlimited bank account, you need to take some time learning where the minefields are in today's VoIP world. Start by reading our Primer on Asterisk Security. We've secured all of your passwords except your root password and your passwd-master password. We're assuming you've put very secure passwords on those accounts as if your phone bill depended upon it. It does! Also read our PBX in a Flash and VPN in a Flash knols. If you're still not asleep, there's loads of additional documentation on the PBX in a Flash documentation web site.

Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX and uncheck the Disable Trunk checkbox. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.

We've also included Trunk configurations for a dozen of our favorite hosting providers to get you started. You can sign up for service with any of them, insert your credentials in the existing trunk, uncheck the Disable Trunk checkbox, and then adjust your outbound route and add an inbound route for your new DID (if you get one).

Stealth AutoAttendant. When incoming calls arrive, the caller is greeted with a welcoming message from Allison which says something like "Thanks for calling. Please hold a moment while I locate someone to take your call." To the caller, it's merely a greeting. To those "in the know," it's actually an AutoAttendant (aka IVR system) that gives you the opportunity to press a button during the message to trigger the running of some application on your Incredible PBX. As configured, the only option that works is 0 which fires up the Nerd Vittles Apps IVR. It's quite easy to add additional features such as voicemail retrieval or DISA for outbound calling. Just edit the MainIVR option in FreePBX under Setup, IVR. Keep in mind that anyone (anywhere in the world) can choose these options. So be extremely careful not to expose your system to security vulnerabilities by making certain that any options you add have very secure passwords! It's your phone bill. 😉

Configuring Email. You're going to want to be notified when updates are available for FreePBX, and you may also want notifications when new voicemails arrive. Everything already is set up for you except actually entering your email notification address. Using a web browser, open the FreePBX GUI by pointing your browser to the IP address of your Incredible PBX. Then click Administration and choose FreePBX. To set your email address for FreePBX updates, go to Setup, General Settings and scroll to the bottom of the screen. To configure emails to notify you of incoming voicemails, go to Setup, Extensions, 701 and scroll to the bottom of the screen. Then follow your nose. Be sure to reload FreePBX when prompted after saving your changes.

A Word About Security. Security matters to us, and it should matter to you. Not only is the safety of your system at stake but also your wallet and the safety of other folks' systems. Our only means of contacting you with security updates is through the RSS Feed that we maintain for the PBX in a Flash project. This feed is prominently displayed in the web GUI which you can access with any browser pointed to the IP address of your server. Check It Daily! Or add our RSS Feed to your favorite RSS Reader. We also recommend you follow @NerdUno on Twitter. We'll keep you entertained and provide immediate notification of security problems that we hear about. Be safe!

Enabling Google Voicemail. Some have requested a way to retain Google's voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you'll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart

;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)

Kicking the Tires. OK. That's enough tutorial for today. Let's play. Using your new softphone, begin your adventure by dialing these extensions:

  • D-E-M-O - Incredible PBX Demo (running on your PBX)
  • 1234*1061 - Nerd Vittles Demo via ISN FreeNum connection to NV
  • 17476009082*1089 - Nerd Vittles Demo via ISN to Google/Gizmo5
  • Z-I-P - Enter a five digit zip code for any U.S. weather report
  • 6-1-1 - Enter a 3-character airport code for any U.S. weather report
  • 5-1-1 - Get the latest news and sports headlines from Yahoo News
  • T-I-D-E - Get today's tides and lunar schedule for any U.S. port
  • F-A-X - Send a fax to an email address of your choice
  • 4-1-2 - 3-character phonebook lookup/dialer with AsteriDex
  • M-A-I-L - Record a message and deliver it to any email address
  • C-O-N-F - Set up a MeetMe Conference on the fly
  • 1-2-3 - Schedule regular/recurring reminder (PW: 12345678)
  • 2-2-2 - ODBC/Timeclock Lookup Demo (Empl No: 12345)
  • 2-2-3 - ODBC/AsteriDex Lookup Demo (Code: AME)
  • Dial *68 - Schedule a hotel-style wakeup call from any extension
  • 1061*1061 - PIAF Support Conference Bridge (Conf#: 1061)
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)

PBX in a Flash SQLite Registry. Last, but not least, we want to introduce you to the new PBX in a Flash Registry which uses SQLite, a zero-configuration SQL-compatible database engine. After logging into your server as root, just type show-registry for a listing of all of the applications, versions, and install dates of everything on your new server. Choosing the A option will generate registry.txt in the /root folder while the other options will let you review the applications by category on the screen. For example, the G option displays all of The Incredible PBX add-ons that have been installed. Here's the complete list of options:

  • A - Write the contents of the registry to registry.txt
  • B - PBX in a Flash install details
  • C - Extra programs install details
  • D - Update-fixes status and details
  • E - RPM install details
  • F - FreePBX modules install details
  • G - Incredible PBX install details
  • Q - Quit this program

And here's a sample from an install we recently completed.


Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. To Philippe Sultan and the rest of the Asterisk development team, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer, Tony Lewis, and the rest of the FreePBX development team, thanks for FreePBX 2.9 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 6 and Asterisk 1.8 and 10 sit up and bark. No one will quite understand what an endeavor that is until they try it themselves. You won't find another CentOS 6 implementation of Asterisk, and Tom has made it look incredibly easy. It wasn't! In fact, when CentOS released 6.1 this week, Tom actually shifted gears (again) and rebuilt PIAF2 (in a couple of days) to take advantage of CentOS 6.1. And, last but not least, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.

Additional Goodies. Be sure to log into your server as root and look through the scripts added in the /root and /root/nv folders. You'll find all sorts of goodies to keep you busy. There's an all-new incrediblefax.sh script that painlessly installs and configures HylaFax and AvantFax for state-of-the-art faxing. The 32-bit install-cepstral script does just what it says. With Allison's Cepstral voice, you'll have the best TTS implementation for Asterisk available. ipscan is a little shell script that will tell you every working IP device on your LAN. trunks.sh tells you all of the Asterisk trunks configured on your system. purgeCIDcache.sh will clean out the CallerID cache in the Asterisk database. convert2gsm.sh shows you how to convert a .wav file to .gsm. munin.pbx will install Munin on your system while awstats.pbx installs AWstats. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. All the other scripts and apps in /root/nv already have been installed for you so don't install them again.

If you've heeded our advice and purchased a PogoPlug, you can link to your home-grown cloud as well. Just add your credentials to /root/pogo-start.sh. Then run the script to enable the PogoPlug Cloud on your server. All of your cloud resources are instantly accessible in /mnt/pogoplug. It's perfect for off-site backups and is included as one of the backup options in the PBX in a Flash backup utilities.

Don't forget to List Yourself in Directory Assistance so everyone can find you by dialing 411. And add your new number to the Do Not Call Registry to block telemarketing calls. Or just call 888-382-1222 from your new number. Enjoy!

Originally published: Thursday, December 15, 2011


VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Safely Interconnecting Asterisk Servers for Free Calling

Adding Skype to The Incredible PBX

Adding Incredible Fax to The Incredible PBX

Adding Incredible Backup... and Restore to The Incredible PBX

Adding Remotes, Preserving Security with The Incredible PBX

Remote Phone Meets Travelin' Man with The Incredible PBX

Continue reading Part II.

Continue reading Part III.

Continue reading Part IV.


Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the PBX in a Flash Forums. It's the best Asterisk tech support site in the business, and it's all free! We maintain a thread with Information, Patches and Bug Fixes for Incredible PBX 2.9. Please have a look. Unlike some forums, ours is extremely friendly and is supported by literally hundreds of Asterisk gurus and thousands of ordinary users just like you. You won't have to wait long for an answer to your question.



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

  1. As part of the yum update process, you'll actually end up with CentOS 6.2 minus the 6.2 kernel. []
  2. If you use the recommended Acer Aspire Revo, be advised that it does NOT include a CD/DVD drive. You will need an external USB CD/DVD drive to load the software. Some of these work with CentOS, and some don't. Most HP and Sony drives work; however, we strongly recommend you purchase an external DVD drive from a merchant that will accept returns, e.g. Best Buy, WalMart, Office Depot, Office Max, Staples. You also can run Incredible PBX 2.9 on a virtual machine such as the free Proxmox server. A security vulnerability has been reported in the Proxmox browser so be sure to run your server behind a secure, hardware-based firewall with no port exposure to the actual Proxmox server from the Internet. []

7 Steps to Skytopia: Pain-Free Calls with Skype and Asterisk

As you probably know, Digium® announced that Skype for Asterisk® would not be available for sale or activation after July 26, 2011. Here we are in November. So what to do? If you're looking for a commercial solution, you're S.O.L. But, if you have a non-commercial PBX for personal use1, then keep reading. We'll walk you through, step-by-step, getting Skype integrated into your PBX in a Flash or Incredible PBX environment. It's easy, but it's a manual process. If you follow the steps below in order, you'll be up and running in about 15 minutes.

Prerequisites. For today's project, we're assuming you have an existing Incredible PBX server running CentOS 5.7. If not, here's our tutorial to get you up and running quickly. You'll also need a keyboard, mouse, and monitor. We strongly recommend a dedicated server such as an Atom-based PC. If you're using a virtual machine, then you'll need a sound card alternative. Try this: /sbin/modprobe snd-dummy.

UPDATE: We've revised this article a bit to accommodate PIAF2 with CentOS 6.2 and Incredible PBX 3. Keep in mind that Skype is a 32-bit application so we strongly recommend a 32-bit platform if reliability matters to you.

Step 1. For inbound Skype calling to work with other implementations including generic PBX in a Flash systems, you'll need to create a SIP URI for your Asterisk server: mothership@127.0.0.1. You do NOT need to expose the SIP port(s) of your Asterisk server to the Internet, and we strongly recommend that you don't! We've previously explained how to set up a SIP URI in this article. The Incredible PBX includes this SIP URI functionality out of the box.

Step 2. You'll also need Java 1.5. To see if it's included in your distribution, issue the following command: rpm -q jdk. If your particular Asterisk distribution doesn't have JAVA 1.5 or higher installed (rpm -q jdk), here's how to do it. Go to the Oracle Technology Network, sign up for a free Oracle web account and log in. While still logged in, accept the binary code license agreement, and click on this link to download jdk-6u12-linux-i586-rpm.bin. Then copy the file to /root on your Asterisk server. Make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin) and then run it. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Whew!

Step 3. You'll also obviously need a dedicated Skype account for your Asterisk server. If you don't have one to spare, download the Skype software for your Mac or Windows PC, and sign up for a free account. You can try out your account by calling our demo hotline: nerdvittles. Get this working on your Mac or PC before proceeding! Then be sure you log out and disable automatic logins on reboot, or you'll have a problem down the road with two machines trying to log in to a single Skype account.

Step 4. Now we're ready to install the remaining software components that your server will need to access Skype. Log into your Asterisk server as root and issue the following commands.

cd /root
mkdir skype
cd skype
wget http://download.skype.com/linux/skype_static-2.1.0.47.tar.bz2
tar jxvf skype_static*
yum -y install xorg-x11-server-Xvfb
yum -y install qt4
yum -y install xterm
yum -y install libXScrnSaver.i386 < == use this for CentOS 5.x #yum -y install libXScrnSaver <== use this for CentOS 6.x wget http://incrediblepbx.com/siptosis.tgz cd .. wget http://incrediblepbx.com/skype-start chmod +x skype-start cp skype-start skype/. cd / tar zxvf /root/skype/siptosis.tgz cd /root/skype

If you'd prefer to avoid all the typing, you can issue the following commands to download a script that will do all the heavy lifting for you. This is for CentOS 5.x systems only:

cd /root
wget http://incrediblepbx.com/skype-setup
chmod +x skype-setup
./skype-setup

For PIAF2 systems running CentOS 6.x, use this instead:

cd /root
wget http://incrediblepbx.com/skype-setup2
chmod +x skype-setup2
./skype-setup2

Step 5. Now there are a few steps to manually configure the software components so that the entire Skype startup process can be automated when your server boots in the future. To begin, you'll need to fire up X-Windows which puts your server in graphics mode. This is the only mode that Skype understands. While logged into your server as root, issue the following command: xinit

NOTE: If xinit won't start on your particular machine, you may need to create /etc/X11/xorg.conf. Here's a generic config file that should work fine for CentOS 5.x systems:

Section "ServerLayout"
Identifier "X.org Configured"
Screen 0 "Screen0" 0 0
EndSection

Section "Device"
Identifier "Card0"
Driver "vesa"
EndSection

Section "Screen"
Identifier "Screen0"
Device "Card0"
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
SubSection "Display"
Viewport 0 0
Depth 16
Modes "800x600"
EndSubSection
EndSection

For PIAF2 users, some have reported issues on Atom machines with seeing a display at all after xinit loads. If this happens to you, don't panic. Simply log into your server from a PC or MAC using SSH. Then run: vncserver :1. Set a password for VNC, and then use a VNC client on your PC or Mac to access VNC at the IP address of your server on display port 1. Now you can continue with Step 6, below.

Step 6. Now we're ready to start up Skype, and get it properly configured. There are two important requirements. First, we want to make sure your credentials are saved for automatic login in the future. And second, we want to configure Skype to run in a minimized state each time it restarts. To begin, click in the white graphics window on your screen using your mouse and issue these commands:

cd /root/skype/skype_static-2.1.0.47
./skype

Click on the Accept button to accept the Skype license agreement. Once Skype loads, enter your Skype Name and Password. Before clicking on Sign In, be sure to check the Automatic Sign In box so that you'll be logged in automatically in the future. Once you're logged in, click on the blue S in the lower left corner of the window to access the Skype Main Menu. Then click Options. When the General tab displays, check the box which says Start Skype minimised in the system tray. Then click the Apply button. To test things out, click on the Sound Device tab and then Make a Test Call. Once you're sure everything is working, click the Close button. Now click on the blue S again and click Quit to shut down Skype.

Step 7. Now we're ready to integrate Skype into the SipToSis middleware so that Asterisk can communicate with Skype. Issue the following commands to start Skype in background mode and then start SipToSis. Be sure to write down the PID for Skype in case we need to kill the app if something goes wrong.

./skype &
cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype. Before clicking Yes, be sure to click the Checkbox to Remember This Selection for future connections! When you click Yes, you'll see the SipToSis CLI indicating that it's waiting for a Skype call.

If you've installed this on an Incredible PBX, Skype should now be functional. From another Skype account, just call the Skype Name that you used to set this up, and your Asterisk extensions should start ringing. To test outbound Skype calling, use an X-Lite softphone connected to an extension on your Asterisk server and dial *echo123 to access Skype's call testing service or *nerdvittles to access our demo.

All that remains is to configure your server to automatically start Skype and SipToSis whenever your system is restarted. Here's how. Press Ctrl-Alt-F2 to get a new login prompt on your server. Log in as root and issue the following command:

echo "/root/skype-start" >> /etc/rc.d/rc.local

Now reboot your server and make sure everything is working.

Navigation Tips. Here are a few navigation tips for managing your Asterisk console on CentOS systems once Skype has been installed:

1. Ctrl-Alt-F2 gets you a new login prompt for your server

2. Ctrl-Alt-F7 gets you back to the SipToSis/Skype session. You can kill SipToSis by holding down Ctrl-C for several seconds. To find the Skype PID: pidof skype. To kill Skype: kill pid#. To restart Skype: skype & and to restart SipToSis, just issue the command again: ./SipToSis_Linux

3. Ctrl-Alt-F9 gets you to the Asterisk CLI.

Setting Up Speed Dials for Skype Friends. One of the wrinkles with Skype is that Skype uses names for its users rather than numbers. If you don't have a SIP URI-capable softphone, there's still an easy way to place calls to your Skype friends using FreePBX®. Just add a Speed Dial number to your FreePBX dialplan. Choose Extension, then select the Custom type, provide an Extension Number which is the Speed Dial number (this could actually spell your friend's name using a TouchTone phone), enter a Display Name for your friend, and add an optional SIP Alias. Then insert the following in the dial field replacing joeschmo with your friend's actual Skype name. Save your entries and reload the dialplan when prompted.
SIP/joeschmo@127.0.0.1:5070

Security Warning. One final note of caution. Do NOT expose UDP port 5070 to the Internet unless you first secure this port with a username and password to avoid Internet intruders using your gateway as a free Skype dialing platform! You do not need 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your Asterisk server so we recommend you keep it securely behind at least a hardware-based firewall.

FreePBX Design. For those not using Incredible PBX, here is the FreePBX setup that Incredible PBX uses and that we recommend. For outbound Skype calls, you have two choices.

1. To place a call to a regular phone number using SkypeOut (which costs you money), you'll simply dial 8 plus the area code and number. Our foreign friends will have to adjust their dialplans and /siptosis/SkypeOutDialingRules.props accordingly. Today's setup assumes 10-digit phone numbers!

2. To place a call to a Skype username using a softphone that supports SIP URI dialing such as X-Lite, you simply precede the Skype username with an asterisk, e.g. *echo123 will connect you to the Skype Call Testing Service or *nerdvittles will connect you to the Nerd Vittles Skype demo.

For incoming Skype calls, the default setup routes those calls to a SIP URI: mothership@127.0.0.1. Whether you point this URI to an extension, ring group, or IVR is up to you. In the default Incredible PBX build, the mothership URI is pointed to the Stealth AutoAttendant, an IVR that plays a welcoming message and then transfers the call to a ring group if no digit is pressed by the caller.

Configuring FreePBX. To put this setup in place, use a web browser to access FreePBX on your Asterisk server. You'll need to create a Custom Trunk and then an Outbound Route.

1. Choose Setup, Add Trunk, Add Custom Trunk. Fill in the form so that it looks like the following using your own CallerID number obviously:

When you're finished, click the Submit Changes button and then reload the dialplan when prompted.

2. Next choose Setup, Outbound Routes, Add Route. Fill in the form so that it looks like this:

When you're finished, click the Submit Changes button. Be sure to move this new OutSkype route to the top position in your Outbound Routes listing in the right margin! Then reload the dialplan when prompted.

3. If you're not using Incredible PBX, add a new DayNight Control 1 option while you're still in FreePBX. Just specify where you want calls routed for Day mode and Night mode. Then, here's the easy way to activate SIP URI support on your Asterisk/FreePBX server. Copy the [from-sip-external] context from the extensions.conf file in /etc/asterisk. Now copy the content into extensions_override_freepbx.conf. Be sure to preserve the context name in brackets! On a FreePBX 2.8 system, make it look like the following. The additions we're making are shown in bold below:

[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => mothership,1,Goto(app-daynight,1,1)
exten => s,n,Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,NoOp(Hangup)
exten => i,1,NoOp(Invalid)
exten => t,1,NoOp(Timeout)

Finally, reload your Asterisk dialplan, and we're finished with Asterisk and FreePBX setup:

asterisk -rx "dialplan reload"

Fedora Builds. For those using recent Fedora builds, these systems have a full implementation of X-Windows and KDE. Just start the system in mode5 (graphics mode), log in, run Skype in one window and start up SipToSis in a terminal window using the commands in Step 7 above. Authorize external use of Skype when prompted.

Where To Go From Here. Well, those are the basics. You now can make one outbound Skype call at a time from your Asterisk server, and you can receive an inbound Skype call on any Asterisk extension when Skype users call your regular Skype name. If you want multiple Skype account support, then you'll need to do some tweaking. What you'll need is the STS Trunk Builder toolkit which is free, but proprietary. Enjoy!

Originally published: Tuesday, November 1, 2011


Great News! Google Plus is available to everyone. Sign up here and circle us. Click these links to view the Asterisk feed or PBX in a Flash feed on Google+.



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...

  1. Excerpt from the Skype Terms of Service: "Subscriptions are for individual use only. Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means). Each subscription is to be used for your own personal communication purposes only, to make calls to another individual. The use of the subscription for commercial gain, such as calling numbers specifically to generate income for yourself or others by placing such calls, is not permitted. Unusual call patterns may be considered indicative of such use and may result in us terminating your subscription and blocking your User Account in accordance with paragraph 11.2." []