Posts tagged: sendmail

Introducing Phone Genie for Asterisk (Email Edition)

From Our Disney Cruise Family Scrapbook Almost two years ago, we introduced Phone Genie for Asterisk®. It let you reconfigure your Asterisk system remotely using your favorite web browser. This included the ability to set and adjust call forwarding, call waiting, and Do Not Disturb for any Asterisk extension. In addition, you could enter Asterisk CLI commands and execute a number of Linux system commands, all from the convenience of your web browser. Phone Genie for Asterisk remains one of the all-time favorite downloads of our readers.

Unfortunately, you don't always have access to a web browser when you're away from your Asterisk server. So today we introduce the perfect complement to the original Phone Genie with our new Email Edition. By following this quick tutorial, you can configure your Asterisk server to respond to any Asterisk CLI command which can be sent from almost any email client on the planet. And we'll perform all this magic with less than a dozen lines of bash scripting. Asterisk CLI commands have almost limitless possibilities. Use Phone Genie to check the status or change the functionality of just about any component on your server.

How It Works. The best way to explain how all of this works is to use a simple example. Let's assume you've left home and forgot to transfer your inbound calls for extension 701 to your cellphone. What we'll do is send a simple email message to a special user account on your Asterisk server that we've set up specifically to handle email directives for your server. Unlike most email addresses, we want this one to be unintuitive so strangers aren't sending messages to your server all the time. Let's assume the address is kxt1498@myserver.dyndns.org for this example. Using any email client, just address a message to that account. For the subject of the message, we'll use the following:

Asterisk: database put CF 701 6781234567

It doesn't really matter whether you include a message with the email. As long as the subject of the email is in the proper form, that's all that matters. The command above activates call forwarding for extension 701 and sends the calls to 6781234567. The command uses standard Asterisk CLI syntax.

On your Asterisk server, we'll have a simple bash script that runs every minute or two to check for new emails in the kxt1498 user's mailbox. If it finds a new message, it will parse the subject line, make certain there is a password match, and then send the command (unaltered) to the Asterisk Command Line Interface for processing. Here's an overview of all the CLI commands. The results of executing the command will be emailed to the address you've configured in the script. This works as both confirmation that your command has been executed and a security alert that your Asterisk system has been accessed using the Email Edition of Phone Genie. In the above example, you would receive an email at the address you've configured in the script with a subject of PhoneGenie. The body of the email would look like this:

Updated database successfully...database put CF 701 6781234567

Prerequisites. This software assumes you are using one of the Asterisk aggregations built on CentOS 5. We've tested it with PBX in a Flash. You'll also need an SMTP server (SendMail or Postfix) that is configured to send and receive emails to and from destinations on the Internet. You do not need a POP3 or IMAP mail server! We've tested this with Asterisk 1.4, but it should work fine with Asterisk 1.6 as well. FreePBX 2.5 or later is required for some functions.

Security Warning. Before we begin, let's pause for a moment to review the enormity of your problems if you do this wrong and to remind you that YOU ARE PROCEEDING AT YOUR OWN RISK. PBX in a Flash in particular is shipped with all outside access to your SMTP server blocked. We've obviously got to remove that layer of security for this software to function properly. But you need to be especially careful with SMTP servers because they can be used to relay SPAM to the entire world if you fiddle with settings that you don't understand. So... DON'T MAKE IMPROVEMENTS THAT AREN'T COVERED HERE UNLESS YOU KNOW WHAT YOU'RE DOING!

This software also gives certain email messages elevated privileges on your Asterisk server so that Asterisk itself can be reconfigured. If you compromise the email account name and password for this application, anybody worldwide can pretty much destroy the functionality of your server. In addition, calls to a certain extension could be rerouted to a very expensive destination on a cruise ship sailing around the world. If your dialplan permitted these calls and you had an account with automatic replenishment from a credit card or bank account, you've got a very expensive problem on your hands. That's one reason that reliable email notification of every Phone Genie transaction is critically important. If you're not getting timely notifications of each Phone Genie transaction, DO NOT USE THIS SOFTWARE until that problem is resolved!

Should you detect that your system has been compromised by receiving an email that indicates a command has been executed on your Asterisk server that you did not initiate, you should immediately disable or remove the script so that no further Phone Genie emails are processed on your server. Be sure to preserve any unprocessed Phone Genie emails for authorities as these may contain important information regarding the source of the emails. These email messages usually are deleted once Phone Genie completes execution of the associated Asterisk commands.

Overview. Here's the drill for today. First, we'll adjust both your hardware- based and IPtables firewalls to allow inbound email delivery to your Asterisk server. Second, we'll remove SendMail from your system and install and configure Postfix to handle the SMTP email chores. This will greatly simplify the security issues in locking down your server from unwanted emails. Depending upon your Internet service provider, installation of Postfix may break outbound email delivery from your server if your provider happens to block outbound traffic on port 25. We'll show you how to fix it. Third, we'll add a new user account on your Asterisk server that will be used exclusively to handle Phone Genie messages. Fourth, you're going to need a fully-qualified domain name for your Asterisk server so that email can be delivered reliably to your server. We'll walk you through getting this set up. Fifth, we'll install and configure the Phone Genie software and run some simple tests to make certain everything is working as it should. Sixth, we'll add the Phone Genie script as a cron job which will be run every couple of minutes to check for incoming Phone Genie emails. Finally, we'll review some of the Asterisk commands that can be executed using the Email Edition of Phone Genie for Asterisk.

Security Design. We've obviously given a great deal of thought to the security issues surrounding this application. The security model we've adopted works like this. First, for an email to get through to your Asterisk server, one and only one email address will work from the Internet. All other inbound email from the Internet will be rejected by Postfix. We strongly suggest you leave it that way. Your email address consists of the special username that we will create on your server plus a (hopefully new) fully-qualified domain name that points to your server. You are well advised to use and keep secret both a non-intuitive and complicated username AND a non-intuitive and complicated, fully-qualified domain name. Only this combination will let the email message through the Postfix filter! Using the correct username and a different FQDN that may also point to your server's correct IP address will nevertheless be rejected by Postfix. The third piece in the security model is the password. If you examine the sample Subject above, you will note that it begins with the word "Asterisk" followed by a colon, a space, and then the Asterisk CLI command. The word "Asterisk" is actually the password, and it can be changed to any password you like. So, if you change your password to FooBaR, then the subject of your message should look like this. Note that the colon followed by a space are also required!

FooBaR: database put CF 701 6781234567

Finally, it should be obvious but... DON'T SEND THESE EMAILS FROM AN UNTRUSTED CLIENT OR A PC IN A PUBLIC PLACE because your email message may get stored in a place that someone else could decipher how to access your server. If you wouldn't leave a $1000 bill beside the computer from which you're sending the email, don't send it! Otherwise, you may lose a good bit more than $1,000. To give you some idea of what's at risk with a compromised system, try sending the following email using your correct email address and password:

FooBaR: help

</sermon>

Firewall Configuration. For purposes of our example today, we're assuming that your Asterisk server is sitting behind a hardware-based firewall/router on a private subnet and that your Asterisk server includes a functioning software-based IPtables Linux firewall. This is the default PBX in a Flash setup that we always recommend. On your hardware-based firewall, you will need to redirect incoming TCP port 25 traffic to TCP port 25 on the private IP address of your Asterisk server. This change often requires a reboot of your firewall/router. Once that change is complete, log into your Asterisk server as root and edit /etc/sysconfig/iptables on PBX in a Flash systems. We need to add a new rule to IPtables which allows incoming TCP port 25 traffic through the firewall. Scroll to the bottom of the file and insert the following lines just above the COMMIT line:

# Allow inbound SMTP traffic on TCP port 25
-A INPUT -p tcp -m tcp --dport 25 -j ACCEPT

Save your additions to the file and then reload IPtables and your network:

service iptables stop
service iptables start
service network restart
service iptables status | grep "tcp dpt:25"

The last command should return an entry from IPtables showing TCP port 25 traffic is now being ACCEPTed into the server. If not, check your entries and repeat the process until this works.

Postfix Installation. Let's continue by removing SendMail from your server and installing Postfix. They both perform the same email functions, but the complexity of SendMail makes the likelihood of a configuration error too risky for us to sleep well. If you understand the intricacies of SendMail and feel comfortable implementing the security model we've described above, by all means, have at it. We'll be happy to share your results with the rest of our user community. In the meantime, here's the Postfix solution. While still logged into your server as root, issue the following commands to uninstall SendMail and install Postfix:

rpm -e --nodeps sendmail
yum -y install postfix

Choosing a Username and FQDN. Before we configure Postfix, you need to decide upon a user account name for your Asterisk server to manage Phone Genie messages. And you also need a fully-qualified domain name which points to the public IP address of your Asterisk server. As mentioned above, we strongly recommend that the username and FQDN be obscure and unguessable. For example, a combination of letters and numbers that don't spell words are good choices. Something like dlrpzh7b3@dhf34.nerdvittles.com will help you sleep well. If you don't have a static IP address and dedicated domain for your server that you can manage, then use an equally obscure FQDN from a provider such as dyndns.org. Something like dhf34.dyndns.org works. You then can configure your Asterisk server to automatically keep your dynamic IP address current. We're going to use these entries as examples below. Obviously, you should choose different entries!

To create the new user account on your server using whatever name you have chosen, here are the commands to issue while still logged into your server as root. Just substitute your chosen username for dlrpzh7b3 in both commands. Be sure to choose a secure password, too.

useradd dlrpzh7b3
passwd dlrpzh7b3

Configuring Postfix. Now let's get Postfix set up for maximum protection. First, move to postfix directory: cd /etc/postfix. Now edit main.cf: nano -w main.cf. Search for the inet_interfaces line in the file: Ctrl-W, inet_interfaces =. Add a hash mark to the beginning of each uncommented inet_interfaces line so that your entries look like this:

#inet_interfaces = $myhostname
#inet_interfaces = $myhostname, localhost
#inet_interfaces = localhost

Next, search for mydestination in the file: Ctrl-W,mydestination =. Comment out each of the lines except the one that looks like this:

mydestination = $myhostname, localhost.$mydomain, localhost

Now add the private IP address of your Asterisk server and your FQDN chosen above to the line so that it looks like this. Don't forget the commas and keep everything on one line.

mydestination = $myhostname, localhost.$mydomain, localhost, 192.168.0.118, dhf34.nerdvittles.com

Finally, move to the last line in the file and make it look like this, all on one line:

smtpd_recipient_restrictions = check_recipient_access hash:/etc/postfix/access, permit_mynetworks, reject_unauth_destination

Save your changes to the file: Ctrl-X, Y, then Enter. Now edit /etc/postfix/access. Move to the very bottom of the file and add two new lines with the following entries using the actual email address and FQDN you chose above instead of the examples. The first line tells Postfix to allow emails addressed to the specified email recipient. The next line tells Postfix to reject all other emails addressed to anyone at this domain. Other domains and public IP addressing are blocked by our mydestination entry above.

dlrpzh7b3@dhf34.nerdvittles.com OK
dhf34.nerdvittles.com REJECT recipient rejected

Save your changes to the file: Ctrl-X, Y, then Enter. Now issue the following two commands:

postmap /etc/postfix/access
service postfix restart

Testing Postfix. Now comes the important part. We need to make sure that outbound emails from your Asterisk server are delivered. And we need to make sure that incoming emails ONLY to the one email address you've designated are received and that all other emails from the Internet are rejected. We can't stress enough how important all three of these tests are. If your Postfix implementation doesn't pass all three, DO NOT PROCEED!

Testing outbound email with Postfix is easy. While logged into your server as root, issue the following command using a destination email address (instead of yourname@gmail.com) where you regularly receive emails:

echo "Hi there" | mail -s Test yourname@gmail.com

Count to 20 and refresh your email's Inbox. If the message is there, you've passed Test #1. If not, check your junk mail folder. If it's still not there, try another email address if you have one. Still no cigar? Then your Internet Service Provider is probably blocking email generated from downstream email servers. For tips on remedying the problem, see this message thread on the PBX in a Flash forums. You might also want to review the Postfix tutorial on dyndns.com. Here's another good tutorial on setting up a Gmail relay using Postfix. And here's another excellent tutorial. Then run the test again until you achieve success.

Testing inbound email to your designated email address is Test #2. Use a web client and send an email message to dlrpzh7b3@dhf34.nerdvittles.com substituting the actual email address you have chosen for your server. Count to 20, log into your server as root and type the following command to retrieve email for user dlrpzh7b3: mail -u dlrpzh7b3. The server should report that you have one new message. Type "d 1" and then "q" to delete the message and quit the mail app. If no email arrives, check the Inbox on your sending client to see if the message bounced and, if so, why. Check your email entries in /etc/postfix/access and /etc/postfix/main.cf for typos and review the steps in Configuring Postfix above. Then repeat the test until you successfully send a message to your designated email address.

Testing inbound email to an unauthorized email address on your Asterisk server is Test #3. For this test, we want to make sure that an email sent to the root account on your server fails. What you'll need for this test is the FQDN that was chosen above. Then, using a mail client, send an email message to root@dhf34.nerdvittles.com using your actual FQDN. Count to 20, log into your server as root, and type: mail. The message you sent should NOT be in the Inbox. Repeat the test by sending a message to root and dlrpzh7b3 @the actual IP address of your Asterisk server. These, too, should both fail. Once you get a passing grade on all three tests, we can move on. The hard part is behind you!

Installing Phone Genie. While logged into your server as root, issue the following commands:

cd /root
wget http://pbxinaflash.net/source/nv/phonegenie.tgz
tar zxvf phonegenie.tgz
rm phonegenie.tgz

Configuring Phone Genie. While still logged into your server as root, edit phonegenie.sh. You will note that there are 3 fields that need to be configured at the top of the file: user, pw, and notify. The user field is the designated user account name that will be used for incoming emails (dlrpzh7b3 in our example). The pw field is the word in every email Subject that precedes the colon, space, and Asterisk CLI command (Asterisk in our example). The notify field is a reliable email address where you regularly receive emails promptly. This is where the results of your Phone Genie email commands will be sent. Choose this email address wisely, as if your bank account depended upon it. It does! Once you have filled in the 3 fields (preserving the quotation marks around each entry), save the file with your changes.

Testing Phone Genie. Now we're ready to try everything out. Using an email client, send an email message to dlrpzh7b3@dhf34.nerdvittles.com (using your actual Phone Genie email name and FQDN). For the Subject, enter the following (substituting the password you created above for Asterisk)... Asterisk: help

After counting to 20, log into your Asterisk server as root and issue the following command:

/root/phonegenie.sh

You should see a display of all of the Asterisk CLI commands and within a minute or so, you should receive an email with the same information at the email address you entered into the notify field in phonegenie.sh in the previous step.

Installing Phone Genie as a Cron Job. Once you have tested several Phone Genie emails manually and you're satisfied that everything is working reliably, you can set up the Phone Genie shell script as a cron job. It should be set to execute every minute or every couple of minutes throughout the day and night. Edit /etc/crontab and insert the command shown below to have the script execute every 2 minutes:

*/2 * * * * root /root/phonegenie.sh > /dev/null

Sample Phone Genie Commands. In addition to all of the traditional Asterisk CLI commands, Phone Genie also supports a number of commands that are specific to FreePBX. These additional commands let you configure call forwarding, call waiting, do not disturb, system speed dials, and blacklist entries on your Asterisk server. For Asterisk CLI command syntax, consult voip-info.org. For FreePBX command syntax, see the listing below. Enjoy!

database put CF 302 8338116666 * Call Forwarding Enable
database del CF 302 * Call Forwarding Disable

database put CFB 302 8238221234 * Call Forwarding on Busy Enable
database del CFB 302 * Call Forwarding on Busy Disable

database put CFU 302 8038445689 * Call Forwarding Unavailable Enable
database del CFU 302 * Call Forwarding Unavailable Disable

database put CW 302 ENABLED * Call Waiting Enable
database del CW 302 * Call Waiting Disable

database put DND 302 YES * Do Not Disturb Enable
database del DND 302 * Do Not Disturb Disable

database put blacklist 6781234567 1 * Blacklist a number
database del blacklist 6781234567 * Remove blacklisted number

database put sysspeeddials 99 6781234567 * Set up Speed Dial 99
database del sysspeeddials 99 * Remove Speed Dial 99
(NOTE: Be sure you enable Feature Code *0 prefix in FreePBX!)

We wish all of you a very Merry Christmas!




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...

A Baker’s Dozen Asterisk Nuggets from the Forums

Whether you’re new to the Open Source VoIP Community or an old-timer, we wonder how many folks actually miss many of the terrific Asterisk® applications that are hidden in message threads on the various Asterisk forums around the globe. In honor of St. Patty’s Day, today we want to take a stroll through the PBX in a Flash forum just to demonstrate what you may be missing by not visiting the forums or subscribing to some of the better syndication feeds.

Skype Gateway to Asterisk. If you read our recent column on integrating a Skype gateway into your Asterisk server, good for you. But, if you missed the forum dialog which followed release of the article, you missed all sorts of enhancements and system integration tips which made the Skype gateway a much better fit on many systems.

CallerID Superfecta. One of the most perplexing issues facing those that implement VoIP telephony solutions is wrestling with CallerID issues which flow from the ongoing Baby Bell phonebook monopoly. Many of you may have tried our CallerID Superfecta application which provides CallerID lookups for FreePBX-based systems using AsteriDex, Google Phonebook, AnyWho, and WhitePages. But, if you’d explored the forum additions to CallerID Superfecta, you would have uncovered an incredibly slick FreePBX installer as well as support for WhoCalled.us and Telcodata plus SugarCRM as well as numerous fixes for syntax changes on the various lookup sources.

Faxing with Asterisk. Other than CalleriD, there’s probably no issue that generates more consternation in the Asterisk community than fax integration. We reintroduced nvfax for Asterisk 1.4 recently. But, if you’d been following the forums, you’d also know that HylaFax and AvantFax now can be easily integrated into PBX in a Flash thanks to the work of Joe Roper and Tony Shiffer.

A2Billing for Asterisk. Another application that’s been difficult to get working with Asterisk has been A2Billing, a sophisticated calling card and PBX billing system. There really never has been a clear, concise cookbook for getting the software installed and properly configured. Once again, thanks to Joe and Tony, this forum thread provides a step-by-step tutorial for getting every facet of A2Billing installed and properly configured.

Asterisk Stickies. This is another promising Asterisk web application for PBX in a Flash that pops up stickies when incoming calls are received. You then can add the contact to your phonebook and also generate the XML code to update the phone directory on Grandstream and Cisco phone sets. It also supports click-to-dial from the web interface. You can keep up with the progress of this developing application in this very active message thread.

Text-to-Speech FreePBX Module. Just today a new TTS module for FreePBX was introduced which lets you generate TTS announcements for use with any FreePBX-based Asterisk system.

Overhead Paging with Asterisk 1.4. Most workplaces need some sort of overhead paging system. With the tips in this thread and any Asterisk 1.4 server, it’s incredibly easy to implement.

Streaming Music on Hold. We introduced streaming audio for Asterisk over three years ago in the Asterisk 1.2 days. A new message thread has updated that technology to support Internet radio using any Asterisk 1.4 server.

Email Alerts on Trunk Failures. For those that rely upon Asterisk systems to do real work, it’s essential to know when access to your carrier has failed so that you can make adjustments to your outbound and inbound trunks. This thread provides a simple tutorial and script to get you started.

Outbound Emails with Asterisk and SendMail. Another one of our Top 5 most perplexing problems with Asterisk is getting an outbound email capability with SendMail working reliably. Part of this is the configuration hassles with SendMail. But service providers such as Comcast have made matters worse by blocking outbound access to port 25 on most non-business accounts. Here’s a message thread that will walk you through configuring SendMail to use Gmail as your outbound SMTP relay host, and you’ll never have an email problem again on your Asterisk server.

Voicemail Notification. Unified messaging may be everyone’s dream but the reality is that it would be nice to be called on your cellphone when a new voicemail arrived at your office. The Voicemail Notification System does just that. And this thread integrates the original design into a FreePBX module.

Configuration Editor for FreePBX. FreePBX stores much of its magic in Asterisk config files. At least in PBX in a Flash, we hide some of these files to protect the integrity of your system. In addition, changes made to some of these files will get overwritten the next time FreePBX is started since it populates a lot of the information in these config files from data stored in MySQL tables. For those that want to learn more about the FreePBX, there now is a configuration file editor which will let you view and edit any FreePBX config file on your system. You’ll find a complete tutorial in the forums.

Hotel-Style Wakeup Calls. A few weeks ago we covered Tony Shiffer’s new add-on module for FreePBX that provides hotel-style wakeup calls for Asterisk systems. This code actually had been available in the forum for several months and is yet another reason to frequently check the new message threads.

Mac OS X Scripting Package. Since publication, a new link to a Treasure Trove of Goodies for Mac OS X has been posted including a link to the new Mac OS X Scripting Package and Asterisk binaries for Mac OS X from Sven Slezak at Mezzo.

Syndication Syntax. Many forums provide a syndication feed link, but many do not. For vBulletin-based forums, the basic syntax for an RSS feed looks like this:

http://fqdn.com/forum/external.php

You can refine the type of feed you want by specifying the type: RSS, RSS2, ATOM, or XML. For example, to pull down a feed from the PBX in a Flash forum, here’s the syntax for the various formats that are supported:

http://pbxinaflash.com/forum/external.php?type=RSS

http://pbxinaflash.com/forum/external.php?type=RSS2

http://pbxinaflash.com/forum/external.php?type=ATOM

http://pbxinaflash.com/forum/external.php?type=XML

You can further refine the feed by narrowing it down to a particular forum of interest. For example, to retrieve the latest threads from the PBX in a Flash Open Discussion forum, the syntax looks like this:

http://pbxinaflash.com/forum/external.php?type=RSS2&forumids=2

Finally, here’s the list of forum ID numbers for the PBX in a Flash forum:

2 – Open Discussion
3 – Help
4 – Endpoints
5 – Trunks
6 – Providers
7 – Wish List
9 – Bug Reporting & Fixes
10 – Add-On Install Instructions

Something We Missed? There are hundreds of additional Asterisk apps hiding in the woodwork. Please share your discoveries by posting a comment and link below. Enjoy!


Want a Bootable PBX in a Flash Drive? Our Atomic Flash bootable USB flash installer for PBX in a Flash has been quite the hit. Special thanks to all of our generous contributors! Atomic Flash provides all of the goodies in the VPN in a Flash system featured last month on Nerd Vittles. You can build a complete turnkey system using almost any current generation PC with a SATA drive and this USB flash installer in less than 15 minutes!

If you’d like to put your name in the hat for a chance to win a free one delivered to your door, just post a comment with your best PBX in a Flash story.1

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

And it still isn’t too late to make a contribution of $50 or more to the PBX in a Flash project and get a free Atomic Flash installer delivered to your door as our special thank you gift. See this Nerd Vittles article for details.


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


Some Recent Nerd Vittles Articles of Interest…

  1. This offer does not extend to those in jurisdictions in which our offer or your participation may be regulated or prohibited by statute or regulation. []

Ringbinder theme by Themocracy