Posts tagged: skype

Skype + Asterisk (still) = Beautiful Music + Free Phone Calls

It's been a disappointing week for Asterisk® with Digium®'s announcement1 that Skype® for Asterisk will no longer be available for sale after July 26, 2011. While many suspected that Microsoft might have been behind the development, it turns out according to Skype2 that this had been in the works for several months so that the company could better focus its attention on Skype for SIP. The problem with that argument, of course, is that at least for now you can't make outbound calls to Skype users with Skype for SIP unless the Skype users happen to be paying for a Skype to Go DID. So where do we go from here?

Well, the good news is that Asterisk 1.8.4.1 now appears to qualify as a stable release with fixes to several nasty bugs which caused some Cisco and Polycom phones to no longer connect. And Skype still produces a client for Linux. And Greg Dorfuss still produces SipToSis, an amazing product that lets Asterisk systems communicate with Skype... in both directions.

The problem has been that the most current release of Skype for Linux required GLIBCXX_3.4.9 which is not part of the CentOS 5.x distribution even though it is available in current releases of Fedora. What you never, ever want to do is mix and match components from one Linux distribution in another. They don't call it Dependency Hell without reason. But, as luck would have it, there's always a guru somewhere that's smart enough to get all the pieces working together.

For those using Incredible PBX, today's your lucky day. We've written a little script that takes all the components outlined above and makes them play nice at least on most Atom-based computers. You still need a sound card that's compatible with CentOS 5.6, but once you're past that hurdle, it's smooth sailing to integrate Skype into your existing system. We'll leave it to you to sort out the licensing issues which do not appear to be problematic if your system is solely for personal use. After all, Skype still produces a Linux client for use on Linux systems, and that's what Incredible PBX happens to run on.

If you use the recommended hardware, today's setup procedure takes less than 10 minutes! Once it's complete, inbound and outbound Skype calling is totally transparent on your Incredible PBX. To reach a Skype number, just dial * plus the user's Skype name from any phone with an alphanumeric keypad. To place a Skype Out call (fees apply), dial 8 plus the user's area code and number. When your 500 million friends on Skype contact you using your Skype name, all of your Incredible PBX phones will ring just like any other inbound call. What's the difference in today's solution and Skype for Asterisk? For openers, today's solution is $66 cheaper. It's free! And, if you're an individual, you won't need Skype's commercial Business Control Panel to make calls. Functionally, the results with your Incredible PBX Skype implementation are identical.3

To make the Skype Magic work, you'll need three pieces of software in addition to The Incredible PBX obviously: Sun's 6u12 Java SE Development Kit, Skype's Static Edition for Linux plus an existing Skype account, and Greg Dorfuss' SipToSis product which manages the Skype Gateway to Asterisk.

As far as hardware is concerned, we're assuming you're using our recommended Acer Aspire Revo to host your Incredible PBX although most Atom-based PCs should work just fine. We don't recommend EEE PCs. With other hardware, your mileage may vary because CentOS 5.6 may or may not support your audio card and graphics mode with your video card. Both are required to get Skype working properly under X-Windows. If you have problems with some other type of hardware, take a look at the tips in our previous article on Setting Up a Skype Gateway to Asterisk as well as the comments.


Installing JDK. Using your favorite browser, go to Sun's 6u12 Java SE Development Kit website, choose Linux for the platform, and agree to the license. Click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to the /root directory of your Incredible PBX. Next, make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin). Then run it: ./jdk-6u12-linux-i586-rpm.bin. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Check to be sure Java was properly installed with this command: rpm -q jdk.

Installing Skype and SipToSis. Now we're ready to load the remaining components. While still logged into your Incredible PBX as root, download and run the skype-setup script. NOTE: We recommend you make a good backup of your system before you begin!

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

Activating Your Skype Gateway. Now we're ready to place your Skype gateway in production. You'll need to perform these steps from the console on your Incredible PBX since we have to run Skype in graphics mode. This may look complicated. It's really not. It's just a bit tedious to figure out the sequence of steps, but we've done that part for you.

WARNING: Be sure that you use a dedicated Skype account on this server! Do not run the same Skype account on any other server or desktop, or it fails!

1. Start up X-Windows: xinit4

2. Start up Skype. While still logged into your server as root, issue the following commands:

skype.sh

Now you need to log in to Skype with your Skype name and password. In this latest version of Skype, we've noticed a quirk. Enter your password before you enter your username, or the system may not accept your username. If the screen appears frozen, press Ctrl-C and try it again. Be sure to set Skype to autologin whenever it is started. Then, in the Skype configuration option, set Skype to always run minimized. Save your settings.

Place a Skype Test Call5 to echo123 to be sure your audio settings are set correctly. Again, with the Aspire Revo, this won't be a problem assuming you have plugged in a microphone and speakers. These can be disconnected after you're sure things are working properly. HINT #2: Intel Atom-based motherboards are a piece o' cake!

Once you've got Skype working and all of the Skype settings configured above, shut down Skype.

3. Restart Skype in Background Mode: skype.sh &

Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. :-) If you forget the PID, you can obtain it with this command: pgrep skype. You can kill Skype with the following command using your actual PID instead of 12345: kill 12345.

4. Start up SipToSis: Press Enter if the command prompt doesn't reappear. Then...

cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype: yes. Important: Be sure to also select the Checkbox to save this setting for future connections!

5. Testing Skype. Go to a softphone (X-Lite recommended!) connected to an extension on your Incredible PBX and dial *echo123. You should be connected to the Skype Call Testing Service. Try *nerdvittles for the Nerd Vittles Demo.

Assuming you have a little money in your Skype Out account, go to any extension connected to your Asterisk server and dial 8 + your home phone number. This will place the outbound call through SkypeOut at 2¢ a minute.

Reboot your server when you're sure everything is working properly.

GUI Tips. Here are a few navigation tips for managing your Asterisk console on your Incredible PBX:

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 decipher your SipToSis PID: pgrep -f SipToSis. To kill SipToSis: kill pid# (that you wrote down). To kill Skype: kill pid# (that you wrote down). To restart Skype: skype.sh & and to restart SipToSis, just issue the command again: ./SipToSis_linux

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

Automating the Skype Gateway Startup. Once everything is working reliably, reboot your server again, log in as root, and issue the command: /root/skype-start. Place a test call again using a softphone on your Incredible PBX. If everything works fine, you now can add the skype-start command to your server's startup script, and you're all set.

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

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. Do NOT expose UDP port 5070 to the Internet by opening a port on your hardware firewall. You do not need UDP 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your server!

Enjoy!

Originally published: Thursday, May 26, 2011



Changes in PBX in a Flash Distribution. In light of the events outlined in our recent Nerd Vittles article and the issues with Asterisk 1.8.4, the PIAF Dev Team has made some changes in our distribution methodology. As many of you know, PBX in a Flash is the only distribution that compiles Asterisk from source code during the install. This has provided us enormous flexibility to distribute new releases with the latest Asterisk code. Unfortunately, Asterisk 1.8 is still a work in progress to put it charitably. We also feel some responsibility to insulate our users from show-stopping Asterisk releases. Going forward, the plan is to reserve the PIAF-Purple default install for the most stable version of Asterisk 1.8. As of June 1, Asterisk 1.8.4.1 is the new PIAF-Purple default install. Other versions of Asterisk 1.8 (newer and older) will be available through a new configuration utility which now is incorporated into the PIAF 1.7.5.6.2 ISO.

Here's how it works. Begin the install of a new PIAF system in the usual way by booting from your USB flash drive and pressing Enter to load the most current version of CentOS 5.6. When the CentOS install finishes, your system will reboot. Accept the license agreement, and choose the PIAF-Purple option to load the latest stable version of Asterisk 1.8. Or exit to the Linux CLI if you want a different version. Log into CentOS as root. Then issue a command like this: piafdl -p beta_1841 (loads Asterisk 1.8.4.1), piafdl -p 184 (loads Asterisk 1.8.4), piafdl -p 1833 (loads Asterisk 1.8.3.3), or piafdl -p 1832 (loads Asterisk 1.8.3.2). If there should ever be an outage on one of the PBX in a Flash mirrors, you can optionally choose a different mirror for the payload download by adding piafdl -c for the .com site, piafdl -d for the .org site, or piafdl -e for the .net site. Then add the payload switch, e.g. piafdl -c -p beta_1841.

Bottom Line: If you use the piafdl utility to choose a particular version of Asterisk 1.8, you are making a conscious decision to accept the consequences of your particular choice. We would have preferred implementation of a testing methodology at Digium before distribution of new Asterisk releases; however, that doesn't appear to be in the cards. So, as new Asterisk 1.8 releases hit the street, they will be made available through the piafdl utility until such time as our PIAF Pioneers independently establish their reliability.




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


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


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


Some Recent Nerd Vittles Articles of Interest...

  1. "Skype for Asterisk will not be available for sale or activation after July 26, 2011. Skype for Asterisk was developed by Digium in cooperation with Skype. It includes proprietary software from Skype that allows Asterisk to join the Skype network as a native client. Skype has decided not to renew the agreement that permits us to package this proprietary software. Therefore Skype for Asterisk sales and activations will cease on July 26, 2011. This change should not affect any existing users of Skype for Asterisk. Representatives of Skype have assured us that they will continue to support and maintain the Skype for Asterisk software for a period of two years thereafter, as specified in the agreement with Digium. We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion. Skype for Asterisk remains for sale and activation until July 26, 2011. Please complete any purchases and activations before that date. Thank you for your business." []
  2. Skype is a trademark of Skype, Inc. "Skype made the decision to retire Skype for Asterisk several months ago, as we have prioritized our focus around implementing the IETF SIP standard in our Skype Connect solution. SIP enjoys the broadest support of any of the available signaling alternatives by business communications equipment vendors, including Digium. By supporting SIP in favor of alternatives, we maximize our resources and continue to reinforce our commitment to delivering Skype on key platforms where we can meet the broadest customer demand." []
  3. Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. []
  4. Starting xinit won't be a problem on the Aspire Revo. But, 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 our purposes:

    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

    []

  5. If the test call fails with a bad audio message, go into Options, Sound Devices and reconfigure your Audio settings until you can place the test call successfully. Otherwise, none of the rest will work! []

The Incredible PBX: Meet the New Kid on the Block

As much as we loved the moniker, the Orgasmatron build was in desperate need of a name change to more accurately describe its true heritage. We didn't look too far for just the right name. Meet The Incredible PBX!

Thanks to the Zero Internet Footprint™ design, it's 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.1 And you won't find a more full-featured Personal Branch Exchange™.

NEWS FLASH: The Incredible PBX is now available for Asterisk 1.8! Go here.

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

The Incredible PBX is much more than just a name change. In addition to all of the Orgasmatron magic including free calling in the U.S. and Canada courtesy of Google Voice, you now get some terrific new features tailored to meet the needs of the individual: randomly generated passwords for all of your extensions, free Skype support and a new backup module both of which we'll introduce over the next few weeks. And CallerID Superfecta now is preconfigured to work out of the box with support from dozens of providers worldwide.

The Incredible PBX Inventory. For those wondering what's included with The Incredible PBX, here's a feature list of components you get in addition to the base install of PBX in a Flash with CentOS 5.4, Asterisk 1.4, FreePBX 2.6, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Please note that A2Billing, Cepstral TTS, Hamachi VPN, and Mondo Backups are optional and may be installed using provided scripts.

Prerequisites. Here's what you'll need to get started:

  • Broadband Internet connection
  • $200 PC3 on which to run The Incredible PBX or a Proxmox VM
  • dLink Router/Firewall. Low Cost: $35 WBR-2310  Best: DGL-4500
  • Free Google Voice account (Available in U.S. without an invite at this link)
  • Free SIPgateOne residential account (U.S. cell to get SMS invite) OR
  • Free IPkall IAX account (recommended for international users)

Installing The Incredible PBX. The installation process is simple and straight-forward. Just don't skip any steps. Here are the 5 Steps to Free Calling, and The Incredible PBX will be ready to receive and make free U.S./Canada calls:

1. Install the latest version of PBX in a Flash
2. Download & run The Incredible PBX installer
3. Set up your two provider accounts
4. Configure a softphone or SIP telephone
5. Run the configure-gv credentials installer

Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. We recommend you install the latest 32-bit version of PBX in a Flash. This new build works much better with newer hardware including Atom-based computers and newer network cards. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.5 operating system. Once installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities. We use virtually identical payloads for all versions of PBX in a Flash.

Download the 32-bit, PIAF 1.6 version from Google, SourceForge, Vitelity, Cybernetic Networks, or AdHoc Electronics. The MD5 checksum for the file is e8a3fc96702d8aa9ecbd2a8afb934d36. Or, if you are feeling really adventurous or if you have new, bleeding edge hardware, try our new 32-bit, PIAF 1.7 build which features CentOS 5.5. This new release is available from SourceForge or Google Docs. The MD5 checksum for the PIAF 1.7 build is 184cdb00142ccdd814b11de23fb00082.

Download the brand-new 32-bit PIAF 1.7.5.5. from SourceForge or one of our download mirrors. Burn the ISO to a CD. Then boot from the installation CD and type ksalt 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! Press Ctrl-C to cancel the install.

On some systems you may get a notice that CentOS can't find the kickstart file. Just tab to OK and press Enter. Don't change the name or location of the kickstart file! This will get you going. Think of it as a CentOS 'feature'. :-)

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, it will reboot. Remove the CD promptly. After the reboot, choose A choose PIAF-Silver option. Have a 10-minute cup of coffee. After installation is complete, the machine will reboot a second time. Log in as root with your new password and execute the following commands:

update-scripts
update-fixes
status

When prompted, change the ARI password to something really obscure. You're never going to use it! You now have a PBX in a Flash base install. On a stand-alone machine, it takes about 30 minutes. On a virtual machine, it takes about half that time. Write down the dynamic IP address assigned to your server after running the status command. You'll need it shortly.

NOTE: So long as your system is safely sitting behind a hardware-based firewall, we do NOT recommend running update-source with The Incredible PBX. The version of Asterisk installed from our payload file is very stable.

Running The Incredible PBX Installer. 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/incrediblepbx.x
chmod +x incrediblepbx.x
./incrediblepbx.x

Have another 15-minute cup of coffee. It's a great time to consider a modest donation to the Nerd Vittles project. You'll find a link at the top of the page. When the installer finishes, READ THE SCREEN!

Here's a short video demonstration of the Incredible PBX installer process:

Either a free SIPgate One residential phone number or an IPkall number is a key component in today’s project. If you are eligible, we strongly recommend a SIPgate One residential account for The Incredible PBX. However, you may elect to use an IPkall account as an alternative. Both are free; however, you cannot register The Incredible PBX to IPkall's servers so you'll need to punch a hole in your firewall to receive incoming calls from Google Voice and IPkall. This step is not necessary with SIPgate accounts since there is a permanent registered connection between The Incredible PBX and SIPgate's servers!

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! Continue reading whichever section below applies to you.

Configuring SIPgate. If you live in the U.S. and have a cellphone, we'd recommend the SIPgate option since no adjustment of your hardware-based firewall is required. Otherwise, skip to the IPkall setup below. Step #1 is to request a SIPgate invite at this link. You'll need to enter your U.S. cellphone number to receive the SMS message with your invitation code. Don't worry. You can erase your cellphone number from your account once it is set up and working properly. Once you receive the invite code, enter it and choose the option to set up a residential account. Next, choose a phone number and write it down. The area code really doesn't matter because Google Voice is the only one that will be calling this number after we get things set up. For now, leave your cellphone number in place so that you can receive your confirmation call from Google Voice in the next step. After that, you'll want to revisit SIPgate and remove all parallel calling numbers. Finally, click on the Settings link and write down your SIP ID and SIP Password. You'll need these in a few minutes to complete the configuration of The Incredible PBX. Now place a call to your new SIPgate number and make certain that your cellphone rings before proceeding.

Configuring IPkall. If you're using IPkall as your intermediate provider, first log in to your hardware-based firewall/router and map UDP port 45694 to the private IP address that you just wrote down. This tells your firewall to pass all IAX2 traffic from the Internet directly to your new server. Don't worry. We have severely restricted which IP addresses can actually send IAX data through the PBX in a Flash IPtables firewall which is an integral part of this build. And, remember, no hardware firewall adjustments are necessary if you're using SIPgate instead of IPkall.

After your firewall is properly configured, you'll need to register for a free IPkall number. This is actually a two-step process. Set it up as a SIP connection when you first register. Then we'll change it to IAX once your new phone number is provided. So your initial IPkall request should look like this:

We recommend area code 425 for your requested number because IPkall appears to have lots of them. If they don't have an available number, your request apparently goes in the bit bucket. You'll know because IPkall typically turns these requests around in a few minutes. Don't worry about the mothership entry. We'll change it shortly. The other issue here is your public IP address. If you have a dedicated IP address, no worries. Just plug in the IP address for SIP Proxy. If it's dynamic, then you'll need to set up a fully-qualified domain name (FQDN) with a provider such as dyndns.com. Once you've got it set up, enter your credentials in the Dynamic DNS tab of your hardware-based firewall to assure that your dynamic IP address is always synchronized with your FQDN. Then enter the FQDN for your SIP Proxy address in the IPkall form. Be sure to make up a VERY secure password. Now send it off and wait for the return email with your new phone number.

When you receive your new phone number, you'll need to revisit the IPkall site and log in with your phone number and the password you chose above. Make the changes shown below using your actual IPkall phone number instead of 4259876543:

It's worth stressing that these settings are extremely important so check your work carefully. Be sure the IAX option is selected. Be sure there are no typos in your two phone number entries. And be sure your FQDN or public IP address is correct. Then save your new settings.

TIP: Be aware that IPkall cancels an assigned phone number after 30 consecutive days of inactivity. If you will be using your number infrequently, it's a good idea to schedule a Weekly Reminder to call the number with a prerecorded message. This will assure that your number stays functional.

Configuring Google Voice. 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. After you've chosen a telephone number, plug in your new SIPgate or IPkall number as the destination for your Google Voice calls and choose Office as the Phone Type.

Google places a test call to your number so you'll have to delay it a bit for IPkall. If you're using SIPgate, go ahead and tell Google to place the test call which will be forwarded to your cellphone. Enter the two-digit code that's displayed when you're prompted to do so. With IPkall, wait until we finish running the credentials configurator below.

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

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.

If you're using SIPgate and you've confirmed your number, revisit SIPgate and remove all parallel calling numbers including your cell number.

Adding Your Credentials to The Incredible PBX. We're ready to insert your credentials and SIPgate/IPkall information into The Incredible PBX. You'll need several pieces of information: your 10-digit Google Voice phone number, your Google Voice account name (which is the email address you used to set up your GV account), your GV password (no spaces!), and your 10-digit SIPgate or IPkall RingBack DID. You'll also need to reenter your passwd-master password which is used to configure CallerID Superfecta. Finally, you'll need to tell the configurator whether you're using a SIPgate or IPkall account. In the case of SIPgate, you'll also be prompted to enter your SIP ID and SIP password. These are NOT the same as your account credentials!!

Log back into your server as root and issue the following command to kick off the configurator: ./configure-gv.x. Check your entries carefully. If you make a typo in entering any of your data, press Ctrl-C to cancel the script and then run it again!! Once you've checked and double-checked your entries, press Enter and The Incredible PBX setup will be completed. You'll need to press Enter again when the script finishes to reboot your PBX. After the reboot, your system will have randomly-generated passwords for every extension and voicemail box that is preconfigured on your system. The DISA password also has been changed. We generate five-digit passwords. If you will sleep better with longer passwords, be our guest. They are easily reset using the FreePBX web interface described elsewhere in this article.

Finally, log back 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 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, and you'll find lots of recommendations on Nerd Vittles. 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.

If you're using SIPgate as your provider with Google Voice, you're ready to place a test call. If you're using IPkall, we still need to verify your IPkall number with Google Voice. Return to Google Voice and tell it to place the test call to your IPkall number which you've already entered as your destination number. Your softphone will ring momentarily. Enter the two-digit code provided by Google Voice, and you're all set.

Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, from another phone, call your Google Voice number. Your softphone should begin ringing shortly. Answer the call and make sure you can send and receive voice on both phones. Hang up. Now let's place an outbound call. Using the softphone, dial your cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. 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. Save the file 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, and 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 this week, we'll point you to some things to play with on your new server. Then, in the subsequent articles below, we'll cover in detail how to customize every application that's been loaded. Nothing beats free when it comes to long distance calls. But nothing lasts forever. So we'd 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 VoIP world is new territory for some of you. Unlike the Ma Bell days, there's really no reason not to have multiple VoIP providers especially for outbound calls. Depending upon where you are calling, calls may be cheaper using different providers for calls to different locations. So we recommend having at least two providers. Visit the PBX in a Flash Forum to get some ideas on choosing alternative providers.

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. 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
  • 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 - PBX in a Flash Support Conference Bridge
  • 882*1061 - VoIP Users Conference every Friday at Noon (EST)



Click above. Enter your name and phone number. Press Connect to begin the call.


Homework. Your homework for this week is to do some exploring. FreePBX is a treasure trove of functionality, and The Incredible PBX adds a bunch of additional options. See if you can find all of them. Also check out Tweet2Dial which uses Twitter to make Google Voice calls, send free SMS messages, and manage your Incredible PBX.

Be sure to log into your server as root and look through the scripts added in the /root/nv folder. You'll find all sorts of goodies to keep you busy. s3cmd.faq tells you how to quickly activate the Amazon S3 Cloud Computing service. And, 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 which we'll cover in a few weeks.

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. Finally, try out the included Stealth AutoAttendant by dialing your own number and pressing 0 while the greeting is played. This will reroute your call to the demo applications option in the IVR.

Originally published: Monday, April 19, 2010

VoIP Virtualization with Incredible PBX: OpenVZ and Cloud Solutions

Adding Skype to The Incredible PBX

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

Adding Multiple Google Voice Trunks 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 the latest 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. So you won't have to wait long for an answer to your questions.

Coming Soon. We haven't forgotten. We'll cover setting up multiple Google Voice accounts for simultaneous calling on multiple channels very soon. And the new (free) Skype Gateway to Asterisk for The Incredible PBX is now available. The FreePBX components already are in place to support inbound and outbound calling via Skype. You can even try a test call to our Aspire One Revo today by dialing nerdvittles from your favorite Skype client. Beginning today, this article will be available on http://IncrediblePBX.com. Then Nerd Vittles will return to our (almost) weekly schedule of new articles. Enjoy!




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


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


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


Some Recent Nerd Vittles Articles of Interest...

  1. Requires a SIPgate One account. []
  2. For Asterisk 1.6 or for 64-bit systems with Asterisk 1.4 or 1.6, use the Cepstral install procedures outlined in this Nerd Vittles article. []
  3. 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 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. []
  4. Mapping a port on your firewall to a private IP address unblocks certain Internet packets and allows them to pass through your firewall directly to an IP device "inside" your firewall for further processing. []

The Incredible PBX: Adding a Free Skype Gateway to Asterisk

Last week we got The Incredible PBX all set up with free worldwide SIP calls, free U.S./Canada PSTN calls using Google Voice with SIPgate or IPkall, and rock-solid Asterisk® security using our new Zero Internet Footprint™ design. Because of licensing restrictions, we couldn't include Skype out of the box. If you're an individual and not a business, today we'll walk you through adding free Skype calling worldwide to your Incredible PBX. With today's addition, the Incredible PBX now provides free calling to nearly a billion phones around the world via Skype, SIP, ENUM, FreeNUM, and U.S./Canada PSTN connections. Yowza!

If you use the recommended hardware, today's setup procedure takes less than 10 minutes! Once it's complete, inbound and outbound Skype calling is totally transparent on your Incredible PBX. To reach a Skype number, just dial * plus the user's Skype name from any phone with an alphanumeric keypad. To place a Skype Out call (fees apply), dial 8 plus the user's area code and number. When your 500 million friends on Skype contact you using your Skype name, all of your Incredible PBX phones will ring just like any other inbound call. What's the difference in today's solution and Digium®'s commercial Skype for Asterisk product? For openers, our solution is $66 cheaper. It's free! And, if you're an individual, you won't need Skype's commercial Business Control Panel to make calls. Functionally, the results with your Incredible PBX Skype implementation are identical.1

To make the Skype Magic work, you'll need three pieces of software in addition to The Incredible PBX obviously: Sun's 6u12 Java SE Development Kit, Skype's Static Edition for Linux plus an existing Skype account, and Greg Dorfuss' SipToSis product which manages the Skype Gateway to Asterisk.

As far as hardware is concerned, we're assuming you're using our recommended $200 Acer Aspire Revo to host your Incredible PBX. With other hardware, your mileage may vary because CentOS 5.4 may or may not support your audio card and graphics mode with your video card. Both are required to get Skype working properly under X-Windows. If you have problems with some other type of hardware, take a look at the tips in our previous article on Setting Up a Skype Gateway to Asterisk as well as the comments. Better yet, visit your neighborhood Best Buy and purchase an Aspire Revo for a hassle-free install.


Installing JDK. Using your favorite browser, go to Sun's 6u12 Java SE Development Kit website, choose Linux for the platform, and agree to the license. Click Continue. Download jdk-6u12-linux-i586-rpm.bin and copy it to the /root directory of your Incredible PBX. Next, make the file executable (chmod +x jdk-6u12-linux-i586-rpm.bin). Then run it: ./jdk-6u12-linux-i586-rpm.bin. Scroll down the wordy license agreement AGAIN and type yes. Java 1.6 then will be installed on your system. Check to be sure Java was properly installed with this command: rpm -q jdk.

Installing Skype and SipToSis. Now we're ready to load the remaining components. While still logged into your Incredible PBX as root, download and run the skype-setup script2:

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

Activating Your Skype Gateway. Now we're ready to place your Skype gateway in production. You'll need to perform these steps from the console on your Incredible PBX since we have to run Skype in graphics mode. This may look complicated. It's really not. It's just a bit tedious to figure out the sequence of steps, but we've done that part for you.

WARNING: Be sure that you use a dedicated Skype account on this server! Do not run the same Skype account on any other server or desktop, or it fails!

1. Start up X-Windows: xinit3

2. Start up Skype. While still logged into your server as root, issue the following commands:

cd /root/skype/skype_static-2.0.0.72
./skype

Now log in to Skype with your Skype name and password. Be sure to set Skype to autologin whenever it is started. Then, in the Skype configuration option, set Skype to always run minimized. Save your settings.

Place a Skype Test Call4 to echo123 to be sure your audio settings are set correctly. Again, with the Aspire Revo, this won't be a problem assuming you have plugged in a microphone and speakers. These can be disconnected after you're sure things are working properly. HINT: Intel Atom-based motherboards are a piece o' cake!

Once you've got Skype working and all of the Skype settings configured above, shut down Skype.

3. Restart Skype in Background Mode: ./skype &

Be sure to write down the PID for Skype in case you need to kill the job if something goes wrong. :-) If you forget the PID, you can obtain it with this command: pgrep skype. You can kill Skype with the following command using your actual PID instead of 12345: kill 12345.

4. Start up SipToSis: Press Enter if the command prompt doesn't reappear. Then...

cd /siptosis
./SipToSis_linux

A message from Skype will pop up asking if you want to authorize external use of Skype: yes. Important: Be sure to select the Checkbox to save this setting for future connections!

5. Testing Skype. Go to a softphone (X-Lite recommended!) connected to an extension on your Incredible PBX and dial *echo123. You should be connected to the Skype Call Testing Service. Try *nerdvittles for the Nerd Vittles Demo.

Assuming you have a little money in your Skype Out account, go to any extension connected to your Asterisk server and dial 8 + your home phone number. This will place the outbound call through SkypeOut at 2¢ a minute.

Reboot your server when you're sure everything is working properly.

GUI Tips. Here are a few navigation tips for managing your Asterisk console on your Incredible PBX:

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 decipher your SipToSis PID: pgrep -f SipToSis. To kill SipToSis: kill pid# (that you wrote down). To kill Skype: kill pid# (that you wrote down). 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.

Automating the Skype Gateway Startup. Once everything is working reliably, reboot your server again, log in as root, and issue the command: /root/skype-start. Place a test call again using a softphone on your Incredible PBX. If everything works fine, you now can add the skype-start command to your server's startup script, and you're all set.

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

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. Do NOT expose UDP port 5070 to the Internet by opening a port on your hardware firewall. You do not need UDP 5070 exposed to the Internet to implement today's gateway solution for inbound or outbound Skype calling from your server!

Enjoy!

Update: As of May 1, you now can set your Google Voice number as your Skype CallerID number. Previously, Google Voice blocked the verification SMS messages, but no longer. Thanks, @zsafwan.

Adding Multiple Google Voice Trunks to The Incredible PBX




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


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


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


Some Recent Nerd Vittles Articles of Interest...

  1. Skype and this suggested implementation are intended for individual use. Your use is, of course, governed by the Skype Terms of Service. []
  2. Here are the actual commands in the skype-setup script if you'd prefer to execute them one at a time:

    cd /root
    mkdir skype
    cd skype
    wget http://www.skype.com/go/getskype-linux-beta-static
    tar jxvf skype_static*
    yum install xorg-x11-server-Xvfb
    yum install qt4
    yum install xterm
    yum install libXScrnSaver.i386
    wget http://pbxinaflash.net/source/skype/siptosis.tgz
    cd /root
    wget http://incrediblepbx.com/skype-start
    chmod +x skype-start
    cp skype-start skype/.
    cd /
    tar zxvf /root/skype/siptosis.tgz
    cd /root


    []

  3. Starting xinit won't be a problem on the Aspire Revo. But, 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 our purposes:

    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

    []

  4. If the test call fails with a bad audio message, go into Options, Sound Devices and reconfigure your Audio settings until you can place the test call successfully. Otherwise, none of the rest will work! []

Nerd Nirvana: Free Google Voice Calling Returns to Asterisk

Lips from Google with Gizmo5In what can only be described as a telephony game changer, Google Voice this past weekend expanded the scope of its offering by providing transparent SIP connectivity through Gizmo5 for inbound and outbound calling. Simply stated, you now can connect virtually any telephone to Google Voice using a garden-variety Internet connection. And the phone can be almost any SIP telephone or a standard home telephone plugged into a $40 ATA. Letting folks make click-to-dial calls through a PC is too geeky for most. But today's offering is a new animal. Google Voice now works with regular telephones.

Did we mention that you get a free phone number of your choice in almost any area code? Did we mention that every call you make throughout the United States and Canada is free? And, believe it or not, transparent Asterisk® support works out of the box as well. If your bread and butter business is SIP termination services in the United States (Are you listening, Vonage and Comcast?), then today probably isn't going to be your lucky day. For everyone else, it may just be remembered as the most important telephony development since the breakup of Ma Bell's monopoly. And now it's clear why Google Voice reserved a million DIDs. They're going to need every one of them... and more! Meet your New Phone Company®, Goliath Google, Inc. What Google Voice was missing was a simple interface to standard telephones, softphones, and SIP. Gizmo5 provides all of those missing pieces... and so much more. How about an almost-free Skype interface for openers.

As many of you know, we were ecstatic when Google Voice arrived with free U.S. calling, voice mail transcription, and SIP connectivity to Asterisk. Our solution lasted less than a week until Google slammed the SIP door and spoiled our party. So we shifted gears and showed you how to use a free Gizmo account and a free Google Voice account to make free SIP calls using Asterisk. Well, that lasted about a week as well although Craig Walker, who founded GrandCentral and now serves as the Google Voice Product Manager, responded to my inquiry about SIP support saying it sounded like a good idea and they would consider it once the initial Google Voice rollout was complete. Guess what? They've kept their promise.

Ironically, we had planned to introduce a new Google Voice solution for Asterisk today and were putting the finishing touches on the article when this news broke over the weekend. We've decided to postpone that discussion because, frankly, the Google Voice-Gizmo5 SIP marriage is the right way to go. It's straight-forward. It's proven technology. It's rock-solid reliable. And it's FREE!

Newly discovered issues with both security and Gizmo5's business model as pertains to making calls through Google Voice have given us pause in recommending the solution described below. In a nutshell, the solution below requires that you provide your Google email credentials to Gizmo5 in order to make the connection to Google Voice for free unlimited 20-minute 3-minute calling. Late yesterday, Gizmo5 announced a new 2¢ per minute fee for Google Voice calling (now described as Gizmo Voice). Yuck!

Even if you don't mind a stranger having unfettered access to your Gmail account, your Google credentials also may be used for other Google services including Google Checkout. Without a clearly defined business relationship between Google and Gizmo5, this would be a huge security risk. Having read several articles which hinted at a business relationship between Google and Gizmo5, we put our security concerns aside. However, when Gizmo5 began changing the ground rules for these calls (almost daily), it raised red flags that Google might not, in fact, be either a business partner or even a willing participant in Gizmo5's creation. As events continued to unfold, we have discovered that Gizmo5 may, in fact, be using a connection process that is not unlike the one we had planned to introduce this week anyway. And we have no business relationship with Google.

Bottom Line: Whether you are using an Asterisk server or not, WAIT! We have an equivalent, secure solution which is now available at no cost. We recommend you disable your Gizmo5-Google Voice setup if you already have put it in place and change your Gmail password! Then read the new Nerd Vittles article for a secure way to connect to Google Voice for free calling.

Our plan today is to show you the easy way to connect Asterisk to Google Voice through Gizmo5 to make free outbound phone calls and to receive free incoming calls. We'll leave the setup for a SIP phone, a generic Asterisk server, and an analog adapter such as the PAP2T-NA for another day. But we'll get to them sooner rather than later.

So, altogether now, welcome back... Googlified Messaging™. Before we begin...

Accounting 101. We hear you asking, "How long can the calls be free?" The short answer is probably not forever but long enough to run just about everyone else out of the business. Beyond that, what we see in our crystal ball pretty much lines up with Tim O'Reilly's talk at OSCON last week. And, at some point, Google may give you a choice of paying for the calls or perhaps volunteering to be their guinea pig for the mother of all indexing experiments. You'd agree to let them record your voice calls without identifying you individually. Then they could transcribe and index all of the keywords in your conversation and use those to identify buying trends, favorite movies, whatever. Remember, you can already say "Pizza" on your iPhone and get a list of nearby pizza parlors so this isn't as far-fetched as you may think. And keep in mind that, in some states, you only need the permission of one party to a telephone conversation to make a recording. Thanks to Amazon, it's been quite a resurgence for Big Brother. We thought we'd join the party with a little Orwellian hypothesizing of our own.

Step #1. If you're starting from scratch, the easiest way to get everything working today including Asterisk is to begin by installing PBX in a Flash, and then run the Orgasmatron Installer. This puts all the pieces in the proper places, and you'll be up and running in under an hour. For the complete soup-to-nuts tutorial, start here.

Step #2. You obviously still need a free Google Voice account to use Google Voice or Google Voice Dialing through Gizmo5. So that's next. If you don't have a Google Voice account, you can request an invite here. Our non-scientific survey suggests that it's taking less than a month to get an invite after you apply. YMMV! Once you have a Google Voice account and a local phone number (Google has reserved a million of them so... not to worry!), then you're all set.

Step #3. Next, you need a Gizmo5 account. If you don't have one, you can sign up for one within FreePBX once you run the Orgasmatron Installer. Or, you can download a Gizmo5 softphone and sign up that way. We're not sure it's required, but be charitable. Put a little money in your Gizmo5 Call Out account. You'll have it for a rainy day or international calling.

Step #4. We'll set up at least one forwarding phone number in your Google Voice account to match your Gizmo5 number. You don't have to actually use it, but it does have to be registered as one of your GV forwarding numbers. Unlike our previous SIP tutorials about Google Voice, you no longer have to configure your Google Voice account to forward all incoming calls to voicemail. As you may recall, this allowed you to call your Google Voice number and press a few keys to make an outbound call instead of listening to your voicemails. With the new Google Voice-Gizmo5 SIP offering, you no longer have to jump through all those hoops. It's a straight SIP-to-SIP-to-SIP connection from your Asterisk server to Gizmo5 to Google Voice.

Step #5. To use Asterisk for incoming calls through Google Voice, you can designate a forwarding number in Google Voice that connects to one or more extensions on your Asterisk system whenever anyone calls your Google Voice number. All you really need for this is one DID. This could be your Gizmo5 number, or it could be a free IPkall or SIPgate DID that's pointed to an extension or ring group on your Asterisk server. Since all of these calls are free, the area code of the DID really doesn't matter. The only number that will really matter to your callers is your main Google Voice number so be sure to select one for your hometown. Incidentally, you can add other forwarding numbers in Google Voice that will ring simultaneously with the DID on your Asterisk server. This could be your vacation home, your cell phone, or even your office phone.

Getting Started. We're going to be jumping back and forth between your Google Voice account, your Gizmo5 account, and the FreePBX web interface to your Asterisk server. So open each account in a separate tab with your web browser. To keep things simple, we're going to assume that you'll be using your Gizmo5 account to connect to your Asterisk server. In Asterisk lingo, the Gizmo5 account looks like any other DID on your Asterisk system.

FreePBX Setup for Gizmo5. If you've run the Orgasmatron Installer, you'll have a new Gizmo5 Integration option under the Setup tab. When you click on that option, you have the choice of either creating a new Gizmo5 account or using your existing account. Fill in the blanks to activate or create your new Gizmo5 account.

Once you've logged in, click Gizmo5 Integration Main Page. Choose Send all calls (except local extensions) through Gizmo5 and click Update Outbound Routes. For the time being, make certain that you have a default inbound route that rings one or more functioning extensions on your Asterisk system. You have to be able to answer an incoming call to complete the next steps. Finally, click on the Outbound Routes option. In the far right column, move the Gizmo5 entry to the top of the list and reload your dialplan when prompted.

If you're using a FreePBX-based system that doesn't have the Gizmo5 Integration option, you'll first need to establish an account at Gizmo5.com by downloading one of the softphones and signing up. After you have completed the sign up process, be sure that you disable automatic startup of the softphone. You can't have your Asterisk system AND the softphone registering to the same Gizmo5 account!

Next, using FreePBX, Add a new Trunk named Gizmo5. For the Peer Details, insert the following using your actual Gizmo5 phone number and password:

type=peer
insecure=very
host=proxy01.sipphone.com
username=1747XXXXXXX
fromuser=1747XXXXXXX
fromdomain=proxy01.sipphone.com
secret=password
context=from-gizmo5-trunk
qualify=yes

Leave the Incoming Settings section blank and then enter the Registration String using your actual Gizmo5 phone number and password:

1747XXXXXXX:password@proxy01.sipphone.com

Save your settings and reload your dialplan when prompted.

Next, create a Default Inbound Route so that calls from Google Voice will be routed to extensions on your server. Then, create an Outbound Route called OutGizmo with NXXNXXXXXX and 1NXXNXXXXXX as the Dial Patterns and Gizmo5 as the main Trunk Sequence . Move this route to the top of your outbound routes to assure that U.S. calls are placed using the Gizmo5 trunk. Reload your dialplan when prompted.

Finally, log into your Asterisk server as root and insert the following lines at the end of extensions_custom.conf in the /etc/asterisk directory. Then reload the dialplan: asterisk -rx "dialplan reload"

[from-gizmo5-trunk]
exten => s,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,Goto(from-trunk,${DID_EXTEN},1)

Google Voice Setup. Log into your Google Voice account and click Settings, Phones, Add Another Phone. This forwarding phone number should be the DID that you want Google Voice to call when you have incoming calls on your Google Voice number. Again, to keep things simple, add your Gizmo5 phone number (747XXXXXXX) and select Gizmo as the Phone Type. You then will be prompted to place a test call and provide a 2-digit number to verify that the number is working. Answer the extension on your Asterisk system when it rings and enter the 2-digit code that's provided.

Gizmo5 Configuration. Log in to your Gizmo5 account using your 1747XXXXXXX account number or username and password. In the new Google Voice section of the form, insert your Google Voice email address and password. This is the email address you used to set up your Google Voice account. Choose "Use for U.S. calls only" and then click SAVE.

July 29 Update. Since this article was released, Gizmo5 has reduced the allowable calling time from unlimited to 20 minutes. Then today it was reduced to 3 minutes. That may be as long as you like to talk on the phone, but it's a major change from what was initially introduced 3 short days ago. Looks like we'll dust off our original article after all. Stay tuned...


Deals of the Week. The nation's premier provider of free directory assistance service, 1-800-FREE-411, now is offering free 5-minute phone calls to most destinations around the world. Just listen to two quick commercials and enjoy your free call. Thanks, @MichiganTelephone. And now you can send free SMS messages worldwide from your iPhone. Thanks, @TruVoIP. Finally, AT&T has the refurbished 8GB iPhone 3G for $49 with a two-year contract.

Originally published: July 26, 2009




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


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


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


Some Recent Nerd Vittles Articles of Interest...

It’s TweedleD: Twitter & SMS Alerts with Every Asterisk Call

Twitter Direct Messages and SMS Instant Messages are great ways to send yourself important notes that you want to keep track of... privately. Today, we'll show you how to harness the power of Twitter and SMS to announce every call on your Asterisk® server with the name and number of the caller as well as the DID of the incoming call. Reconfiguring your Asterisk server takes less than 10 minutes. We think you'll find this to be the perfect complement to our free Urang II desktop screenpop utility.

Prerequisites. You'll need one of the Asterisk aggregations to get all of this working quickly. At a minimum, you need Asterisk 1.4 or 1.6, FreePBX, Apache, PHP 5, PHP/AGI, and SendMail or an equivalent. You'll also need a service that provides CallerID name lookups such as CallerID Superfecta. We're assuming you already have all of these components working including outbound email which is necessary to deliver the SMS alerts.

Overview. The design strategy for TweedleD is pretty straight-forward. When a call comes into your server, FreePBX should catch the call in an inbound route that looks for incoming calls on a particular DID. A CallerID Lookup source will be used in the Inbound Route to look up the name of the caller based upon their CallerID number. Then we will use a Custom Destination to route the call to a special Asterisk context which will run the TweedleD AGI script. This script actually sends out the Twitter Direct Messages as well as the SMS messages to your cellphone. Once the script completes its work, the context will send the call along to its final destination, e.g. an extension, a ring group, a day/night control, or an IVR.

NOTE: There is a 100-150 API calls per hour per IP address limit with the Twitter API. If you also are using one or more Twitter clients at your site, this is something worth keeping in mind if you have a busy phone system.

1. FreePBX Custom Context. The trick with designing FreePBX dialplans is to build them in reverse. So we'll start with the custom context and work our way back to the inbound route. First, log into your server as root and edit extensions_custom.conf in /etc/asterisk. At the end of the file, we want to create a new context that looks like this. We've provided several sample destinations at s,5 to show you the syntax. The first routes a call to an extension or ring group number. The second routes the call to Day/Night Control #1. And the third routes the call to an IVR. To find the correct IVR number or day/night control for what you want to do, review the IVR and app-daynight contexts in extensions_additional.conf to find the one you need. Obviously, you only use one s,5 line. Comment out the remaining ones as shown below. For Asterisk 1.6, replace the vertical bars in line s,3 with commas.

[custom-twitter]
exten=>s,1,Answer
exten=>s,2,Wait(1)
exten=>s,3,AGI(nv-twitter.php|${CALLERID(name)}|${CALLERID(num)})
exten=>s,4,NoOp(${CUSTOMDATA})
exten=>s,5,Dial(local/701@from-internal)
;exten=>s,5,Goto(app-daynight,1,1)
;exten=>s,5,Goto(ivr-6,s,1)
exten=>s,6,Hangup

2. Installing PHP/AGI Scripts. Next we need to install the script that actually generates the messages for Twitter and SMS. In addition, we'll install Justin Poliey's terrific twitter.lib.php which is a PHP implementation of the Twitter API that lets you do just about anything with Twitter that you could do with the Twitter API itself. For a good writeup on the capabilities of twitter.lib.php, see Antonio Lupetti's article. To install the necessary code, issue the following commands while still logged in as root:

cd /
wget http://bestof.nerdvittles.com/applications/TweedleD.tgz
tar zxvf TweedleD.tgz
rm TweedleD.tgz
cd /var/lib/asterisk/agi-bin

3. Configuring TweedleD. Now we need to configure the PHP script with your Twitter and SMS credentials so that TweedleD knows where to send the messages. Edit nv-twitter.php in the /var/lib/asterisk/agi-bin directory on your server: nano -w nv-twitter.php. You'll see a section of code near the top of the application that looks like this:

//-------- DON'T CHANGE ANYTHING ABOVE THIS LINE ----------------

$tweet = 0;
$username = "your-twitter-name";
$password = "your-twitter-password";
$user4msg = "recipients-twitter-name";

$sms = 0;
// $smsaddress = "1234567890@txt.att.net" ; // AT&T
// $smsaddress = "1234567890@message.alltel.com" ; // AllTel
// $smsaddress = "1234567890@messaging.nextel.com" ; // Nextel
// $smsaddress = "1234567890@messaging.sprintpcs.com" ; // Sprint
// $smsaddress = "1234567890@tmomail.net" ; // T-Mobile
// $smsaddress = "1234567890@vtext.com" ; // Verizon

$debug = 0;
$newlogeachdebug = 1;

$emaildebuglog = 0;
$email = "yourname@yourdomain" ;

//-------- DON'T CHANGE ANYTHING BELOW THIS LINE ----------------

There are four things you can enable and, depending upon what choices you make, you need to add your credentials for the various options. Let's go through them in the order they appear.

To enable Twitter Direct Messages, change $tweet=0 to $tweet=1. $username and $password are used to set your Twitter login credentials for your Twitter account. You must have one! $user4msg is the Twitter account name where the DMs should be delivered. They do not necessarily have to be delivered to your Twitter account although they can be. Be sure to preserve the quotes! And, remember, you can only send direct messages to people on Twitter that you are following, and they must be following you as well. If in doubt, attempt to send a direct message in Twitter to the desired recipient. Then you'll know for sure. TweedleD provides no error messages if things don't work. :-(

To enable delivery of SMS messages to your cellphone, set $sms=1. TweedleD actually delivers SMS messages using email. Virtually all of the carriers provide an SMS-Email Gateway for this purpose. The trick is knowing the email domain for your desired carrier. The full list is available here. And here is an even newer Email to SMS Gateway list. In the script, we've provided samples for the major U.S. carriers so, if yours is in the list, just uncomment the appropriate line by removing the leading // on the line and replace 1234567890 with your 10-digit cellphone number.

If you want a debug log generated for each call, set $debug=1. The default is to overwrite the previous log (/var/log/asterisk/nv-twitter.txt) with each new call. If you'd prefer an ever-growing log, set $newlogeachdebug=0.

The debug log also can be emailed to you. Set $emaildebuglog=1 and enter your email address in $email.

When you've completed the configuration, save the file (Ctrl-X, Y, Enter) and reload the Asterisk dialplan: asterisk -rx "dialplan reload".

4. FreePBX Setup. Finally, we need to configure FreePBX to support TweedleD. We'll add a Custom Destination and then adjust our inbound route(s) to support the custom context we added in Step 1. Open FreePBX with a web browser and choose Admin, Tools, Custom Destinations, Add Custom Destination. For the Description, call it TweedleD. For the Custom Destination, enter the following: custom-twitter,s,1. Then click Submit. Now edit your Inbound Routes for the DIDs for which you wish to activate TweedleD. For each inbound route, make sure you have activated a CallerID lookup source and then set the destination to: Custom Destination: TweedleD. Save your changes and reload the dialplan.

Trying Out TweedleD. Now you're ready to place a test call. Just call into your system on one of the DIDs that has been configured for TweedleD. On the Asterisk CLI, you should see an entry that identifies the CallerID name and number of the caller. And, if you've activated delivery of notifications to Twitter and/or SMS, the messages will magically appear within seconds. Enjoy!


Tips of the Week. If you hurry, you can get 25GB of free, password-protected online storage with the new Micro$oft SkyDrive offering. Or, for a free incoming call number for your Skype account, check this service out. A new precompiled version of Asterisk 1.6 for Mac OS X Leopard has been released. You can download it here. Finally, Sprint is offering a Netbook for 99¢ with any 2-year EVDO contract at Best Buy stores this week. Regular price for the Compaq Mini is $389.99. Add your cell phone or Asterisk DID to the DO NOT CALL Registry. Just call 888-382-1222 using the CallerID of the number to wish to block.


Sign of the Times. Before you get too comfortable with all your free Google apps, you might want to read this article. Despite assurances to the contrary from Google's President of Enterprise, it seems that the free version of Google Apps has quietly disappeared.

Update: Google Apps team invokes Brain Fart defense saying they momentarily forgot about "free" but it won't happen again. Heh, heh. Official Blog Posting.


VoIP Users Conference. Come join the fun. The VoIP Users Conference is held every Friday at noon, Eastern time. ISN: 8647*1061 (Hint: 8647 spells VOIP on your phone).


The World According to Twitter. Be on the lookout for David Pogue's new book, The World According to Twitter. It goes to press today! Incidentally, Pogue's books are just about the only reasonably priced tech books available for the Kindle, and they're all excellent.

Tomorrow's the Big Day. The last time this happened, it was a pretty quiet year. Just after noon tomorrow comes your once-in-a-lifetime moment when the time will officially be 12:34:56 7/8/9. The last time we had this much fun in our lifetime was... uh, about a year ago: 01:23:45 6/7/8. Can't wait for next year's thriller: 23:45:67 8/9/10. Ooops! That won't work on most clocks. Will it? Amazing what you can learn on Twitter. Isn't it? Thanks, @ejovi and @ev.




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


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


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


Some Recent Nerd Vittles Articles of Interest...

Google Voice: Is the SIP and Asterisk Honeymoon Over?

Lips from Google“Well. That was quick.” Not encouraging words to hear from your new best friend. Google doesn’t make many mistakes so let’s give their decision to shut down SIP connectivity to Google Voice a little more time to percolate before concluding that they’ve thrown the baby out with the bathwater. The knee-jerk reaction is simply to write off Google as having about as much technical and business savvy in the VoIP market as AOL demonstrated… twice. But that’s not the Google many of us have known and done business with. And it’s the antithesis of everything Google Android and the company have sought to promote.

Update: The original SIP interface to Google Voice described in this posting no longer works. A new approach that really works is now available on Nerd Vittles at this link.

For the record, let’s back up a minute and review what transpired. Last Monday we (and others) released a tutorial showing users how to almost transparently connect Google Voice to Asterisk® PBXs as either a SIP extension or a trunk. The beauty of this was that it added a great new, low-cost telephony provider to the worldwide mix. The short-term advantage to Asterisk users was that calls within the U.S. currently were free although Google already has announced that those darn “accountants” have told them that they’re going to be forced to charge for the service one day soon. Cough cough!

In the process of testing this SIP connectivity, what we discovered was the only layer of protection standing between your wallet and free worldwide phone calls for every creep on the planet was a 4-digit PIN. That translates into 10,000 SIP calls to break into any user’s account. Even without the assistance of BOTs, that afforded your shiny new Google Voice account less than an hour of protection with a well-written SIP dialer and no added protection from Google Voice. By Friday, Google had closed the hole and blocked all SIP connectivity except for Gizmo.

The simple solution to open up safe SIP connectivity to Google Voice would be the addition of either an IP address field or a SIP URI in the Google Voice configuration options. SIP calls to and from that address would be allowed. All other calls would be blocked.

And why is this a good idea? First, it promotes the SIP open source standard. See Andy Abramson’s blog for a thought-provoking analysis of where this could ultimately lead. Second, it brings Google Voice connectivity to an enormous pool of users most of whom are tech-savvy and influential in the VoIP marketplace. Millions of Asterisk systems already have been deployed worldwide. Third, it’s the right business decision. Can you spell S-K-Y-P-E? At a time when Skype is opening up its network to SIP connectivity through Skype for SIP and Skype for Asterisk not to mention corded and cordless telephones, what possible business case could be made for introduction of a closed-platform VoIP service with no outside connectivity except through MaBell landlines? Hello!

This may come as a shocker to the Google accountants, but the call pricing and the double-hoop outbound dialing through Click2Dial aren’t that great. Comparable SIP call pricing is available from thousands of providers worldwide. And voice transcription through the Click2Dial voicemail service is downright horrendous. We proved that quickly with our Google Voice demo system.

It comes down to this. The one truly distinguishing factor with Google Voice is Google. At a time when Google has been at the forefront of open source telephony in the cellphone space with Android, the current Google Voice design is a giant step backwards. Rumor has it that Ma Bell had an offering that rang phones in multiple locations about 70 years ago. It was called a Party Line. How are they doing with that? We hope Google does the right thing and opens its new service to safe SIP connectivity. It’s the right and the bright thing to do.

The Honeymoon Ain’t Over… The Return of Googlified Messaging With Free U.S. Calling


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


Some Recent Nerd Vittles Articles of Interest…

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