We're going to take a serious look at Asterisk® through the eyes of a typical new user today. Our objective is to turn newly built Asterisk servers into stellar performers, IP telephony systems that work reliably without the quirks that are all too familiar to those of us who have tiptoed through the minefield for many years. Whether you've chosen to run PBX in a Flash, or a trixbox system, or Elastix, or rolled your own Asterisk system, that's the least of your problems. And it doesn't really matter which flavor you chose because most of the pitfalls we'll be discussing today apply more or less to all of the distributions. Our yardstick for whether your system is performing satisfactorily is straightforward. When your significant other begins screaming for the return of a plain old telephone, you know, one where people on the other end of a call can actually hear what you're saying... you've got a problem.

Download Blues. You can't build an Asterisk-based turnkey system without knowing how to deal with an ISO download. If you have questions about how to create a usable CD from an ISO download or, if your newly minted CD won't boot, follow these simple steps. With a Mac, use Roxio Toast. Choose Copy, click Image File, and drag the ISO file you downloaded into the folder. Click Burn after inserting a blank CD. If you don’t own Toast for your Mac, go to the Applications->Utilities folder and run Disk Utility. Click on Images->Burn from the Title Bar and choose the ISO file you downloaded. Then click Burn to begin. For those in the PC World, you’ll need either Roxio Easy CD Creator or Nero to create a CD from an ISO image. With Easy CD Creator, choose Create Data CD. Then in the File menu, select Create CD from Image, and choose your downloaded file. Now click burn to begin. With Nero, go to Recorder from the top menu and choose Burn Image. Select your download file. Then from the Burn Compilation Window, choose Burn to begin.

Hardware Nightmare. Our Wild Ass Guess (WAG) would be that 90% of the installation problems experienced by new Asterisk users are directly related to crappy hardware. If it sounds like we're tired of hearing about this, you'd be right. The issues range from clone X100P cards that don't work (those that do work usually don't work for long!) to 10 year old systems that barely work to $3,000 top-of-the-line dual everything systems that Linux simply does not yet recognize because the hardware is so new that the glue isn't even dry on the motherboard. The video card is brand new, the onboard network adapter has been in production less than a month, and the SATA RAID drive adapter has been customized just for Dell. Guess what, Dude? The operating system won't load. ATTN: Everybody. Do yourself (and us) a favor. Throw your 10-year-old system in the recycle bin where it belongs. And don't replace it with the most expensive new system from Dell that you can find. We've got nothing against Dell by the way. Keep in mind that we're not loading Windows Vista Premium Deluxe that needs 10,000 horsepower to get out of bed every morning. For a Linux-based telephony server that is going to support under 100 people, the $3,000 server is just overkill and will cause many more problems than it solves. Instead, scratch together $200 and buy yourself a new WalMart Special, a.k.a. the Everex Green PC. You also can get one from NewEgg if you hate WalMart.everything. Now add a gig of RAM for $25 and call it a day. Bottom line: It works. It's reliable. It's new. And it's got performance to spare. Worried about a system failure? Then buy two of them, and we'll show you how to build mirrored servers in coming weeks.

Hardware Nightmare, Part II. For newbies that skimp on hardware, their next purchase is usually the cheapest SIP telephone on the planet. Don't! It's a Death Wish Come True. A week later you'll be wondering why all your friends say it sounds like you're calling from a tunnel. The Little Mrs., of course, has long since begun making all of her calls on a cellphone... which tells you how bad your new system really is! Our advice: Take the $200 you saved buying the WalMart Special above, and buy yourself ONE decent SIP telephone. You'll never be sorry. The Aastra 57i is a perfect phone, period. You can read why here. We even have free software that will automatically configure Aastra 57i's for you. All you have to do is plug it in. And, if you like the flexibility that comes with cordless handsets, splurge for the 57i CT for about $100 more, and you'll have the best phone plus one or more cordless handsets with incredible range.

Software Nightmare. Whether you barely understand Linux or consider yourself a Linux guru, unless you know just as much about Asterisk, save yourself (and the existing Asterisk community) weeks and weeks of headaches. Download one of the Asterisk aggregations that's already been built for you such as PBX in a Flash. In the case of PBX in a Flash, it includes all of the source code necessary to recompile anything on the system once you get your feet wet. Believe it or not, the people that put these aggregations together have decades of Linux, networking, and telephony experience. They actually know what they're doing (in most cases), and the FreePBX web interface to Asterisk that is included in most of these packages was written by some of the best Asterisk gurus on the planet. These aggregations are self-contained ISO images that include the operating system and every piece of the puzzle that you'll need to get an Asterisk system up and running in under an hour. No small feat! If you pick the right one, everything works out of the box, and you can keep it current by issuing one simple command from the Linux prompt... any time you like. It's also easy to add your own pieces down the road using the included compiler and compilation tools. For those that say "I wanna learn as I go" but don't know the difference in a Dialplan, a Bedpan, and a Portapotty (HINT: see inset), here's a tip. Start with an aggregation and then build your own Asterisk system from the ground up... in about six months after you return from Asterisk Bootcamp. In the meantime, pick up a copy of Linux for Dummies. If you're too cheap to cough up the twenty bucks, at least read Joe Roper's Conversational Linux for Newbies. It's free.

It's Your Firewall, Stupid. I wish I had a nickel for every message thread that has been written that goes something like this. "I can make calls out of my system, but the people I call can't hear me." Or vice versa. The answer is pretty simple if you stop and think about it for a second. A phone call has two participants. One talks and the other one listens. Then you take turns. At least that's the theory. For that to actually work in the world of Internet telephony, the talking legs of the call have to be able to get from Point A to Point B and from Point B to Point A. If your IP-based telephone or Asterisk system is sitting behind a firewall/router, you have to configure your router to pass the incoming data into the server and telephone on your private network. If the telephone or Asterisk system on the other end of the call happens to also be sitting behind a firewall/router, then we have what's called "double NAT issues." And, no, this doesn't refer to no-see-ums on a steamy summer night in Dixie. Bottom line: If any of this communications traffic can't find it's way to the other end, then someone can't hear all or part of the telephone conversation.

To fix NAT problems with Asterisk, you simply tell your router to forward all data received on UDP ports 4569, 5004 to 5037, 5039 to 5082, and 10000 to 20000 to the private IP address of your Asterisk server. You also must make certain that the following entries exist in /etc/asterisk/rtp.conf:

[general]
rtpstart=10000
rtpend=20000


And bindport = 5060 must exist in the [general] context of /etc/asterisk/sip.conf. The aggregations take care of the rtp.conf and sip.conf setups for you. But you must reconfigure your router/firewall. Last, but not least, you probably need to complete the next step below as well.

Wherefore Art Thou, Server? If you plan to add additional telephones to your system which are not behind the firewall with your Asterisk server, then those phones have to know the public IP address of your server... all the time. The same holds true with some Internet telephony hosting providers. In lieu of a static IP address, you can use a fully-qualified domain name, e.g. mypbx.dyndns.org. This avoids a problem if your Internet service provider only gives you a dynamic IP address which changes from time to time. There's one more step in making this work. You have to set this information up in Asterisk. Here's how.

Log into your Asterisk server as root and edit sip_custom.conf: nano -w /etc/asterisk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:

externip=180.12.12.12
localnet=192.168.1.0/255.255.255.0      (NOTE: The first 3 octets need to match your private IP addresses!)

If you have a public address that changes and you're using DDNS, then the settings would look something like the following:

externhost=mypbx.dyndns.org
localnet=192.168.0.0/255.255.255.0      (NOTE: The first 3 octets need to match your private IP addresses!)

Once you've made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.

Be aware that, with some hosting providers, you may experience problems with the externhost approach outlined above. If your ISP only gives you a dynamic IP address, you still can use the externip approach above so long as you have a method to frequently verify your IP address. The approach we actually use on our network is to run a little script every 5 minutes. If it finds that your outside IP address has changed, it will automatically update your sip_custom.conf file with the new address. To use this approach, create a file in /var/lib/asterisk/agi-bin named ip.sh. For this to work, you have to be able to ping your fully-qualified domain name and get a response! Here's the code:¹

#!/bin/bash
fqdn="mypbx.dyndns.org"
localnet="192.168.0.0"
externip=`ping -c 1 $fqdn | cut -f 2 -d "(" | cut -f 1 -d ")" -s ↩
| grep -m 1 ^`
if [ -e /tmp/$externip ] ; then
echo No IP Update Required ;
else
echo IP Update Required ;
touch /tmp/$externip ;
echo "externip=$externip" > /etc/asterisk/sip_custom.conf
echo "localnet=$localnet/255.255.255.0" >> /etc/asterisk/sip_custom.conf
asterisk -rx "dialplan reload" ;
fi

On line 2 of the above code, enter the fully-qualified domain name for your server that is registered with your DDNS host. Take a look at this thread for information on DNS-O-Matic which is free.

On line 3, enter the internal subnet for your server. This is usually 192.168.0.0 or 192.168.1.0. YMMV!

Save the file and give it execute permissions: chmod +x /var/lib/asterisk/agi-bin/ip.sh. Then make asterisk the file owner: chown asterisk:asterisk /var/lib/asterisk/agi-bin/ip.sh.

Finally, add the following entry to the bottom of /etc/crontab:

*/5 * * * * asterisk /var/lib/asterisk/agi-bin/ip.sh > /dev/null


Snap, Crackle, and Pop. No. Your phone calls are not supposed to sound like a bowl of Kellogg's Rice Krispies. If they do, it usually means your Internet bandwidth is insufficient to support a reliable VoIP call. Using an uncompressed codec such as ULAW, a single call requires roughly 128 kbps of bandwidth in both directions for a reliable conversation. A full T1 can handle roughly 20 simultaneous calls. If you have a dial up Internet connection, do your friends a favor. Go back to tin cans and a string. It'll work just as well and maybe better. Keep in mind that most ISPs do not offer any QOS guarantees with their service and upstream bandwidth is severely restricted. Not surprisingly, this seems to have gotten worse as more and more ISPs try to steer their customers towards their own VoIP offerings. If you have Internet bandwidth to spare but have a busy LAN, you may want to consider a router that provides increased throughput for certain types of data, e.g. SIP and IAX traffic. Most gaming routers provide good traffic shaping functionality. For example, the dLink DGL-4300 Gaming Router provides excellent results and is currently available at Amazon for under $85 after rebate. Another option is to use a different codec for your calls. See this table for the bandwidth calculations. But be aware that as VoIP data gets compressed, you also run the risk of serious degradation in calls if there is any appreciable packet loss because of the geometric effect this has on compressed data. See this thread for some other troubleshooting tips.

Got Those Disappearing Email Blues. Where did my emails go? Nowhere is the usual answer. Sending email messages with your latest voicemails attached is a wonderful feature that PBX in a Flash and other FreePBX-based systems fully support. There are two common problems in sending emails from your LAMP-based Asterisk server. Either your server isn't configured to send out email or your ISP is blocking the transmission of emails that originate from your system. It's usually easier to troubleshoot email problems by first determining whether your ISP is blocking the emails. Then it's pretty simple to test whether your server is properly configured to send the messages... but, first, a brief history lesson.

Many ISPs don't like downstream servers that function as so-called SMTP hosts because of SPAM and email relay hosts. An improperly configured SendMail server can be used to generate thousands of messages an hour from anyone with an Internet connection. One of the first SPAM messages we received after creation of the Department of Homeland Security was a message using a DHS sendmail server as an email relay host. That inspired confidence. To avoid this problem, ISPs do several things. Typically they block port 25 on their servers so that you can't send out email from downstram SMTP servers. Instead, you have to use their SMTP server to send all outbound email. Comcast takes it a step further. On some systems, they block port 25 on your cable modem so that email never leaves your home or office. Do they typically tell you when they do this? Of course not. While all of this is done in the name of reducing SPAM, it's also a convenient excuse for imposing service restrictions which also happen to save them bandwidth... which you are paying for.

To test whether your ISP is blocking port 25, log into your Asterisk server as root and issue the following command:

telnet nerdvittles.com 25

If your provider isn't blocking port 25, you should get a response like this:

Trying 69.89.21.89...
Connected to nerdvittles.com (69.89.21.89).
Escape character is '^]'
220-We do not authorize the use of this system to transport unsolicited,
220 and/or bulk e-mail.


If your ISP is blocking port 25, then the first step to get email flowing from your Asterisk server is to reconfigure SendMail in one of two ways. You can either send the messages through your ISP's SMTP server (and this won't work if port 25 is blocked on your cable modem!) or you can send secure messages using gMail as your SMTP relay host on port 587. This requires that you set up a free gMail account first. For detailed instructions on the gMail setup, go to this message thread and follow the instructions. For an example of using Comcast as your SMTP relay host using port 587, read this thread.

Now we're ready to configure your Asterisk server to reliably send out email messages. There's a simple trick to get this working. A fully-qualified domain name for your server must match the "from" address for the email messages that are sent. This domain does not actually have to be accurate so long as you don't expect to get return emails. Think of it as putting a fake return address on a letter which you mail. It doesn't keep the letter from getting to the designated destination. It just means that you'll never get it back if it were incorrectly addressed. So... our recommended scenario is to do the following. Log into your server as root and edit /etc/hosts. Insert pbxinaflash.dyndns.org in front of pbx.local and separate the entries with a space. Save the file and then restart your network: service network restart. Now edit /etc/asterisk/vm_general.inc and change the serveremail line to read as follows: serveremail=vm@pbxinaflash.dyndns.org. Save the change and reload your dialplan: asterisk -rx "dialplan reload".

Finally, we want to send a test message to be sure everything works. Then you can use FreePBX to tell Asterisk to deliver voicemails to your email address by editing your Extensions settings. To send a test message, log into your server as root and type the following using your real email address. Wait a minute and then check your mailbox (including your SPAM mailbox) to be sure you got it somewhere.

echo "test" | mail -s testmessage nerduno@dyndns.org


Decipherable TouchTones Really Are Part Magic. For the poor soul that finally has a system where he can both speak and hear on the phone (just like in the Old Days), the next hurdle usually rears its head the first time you connect to your favorite doctor's office or credit card company and need to press zero for customer service. After pressing 0 for the hundredth time, you conclude that the buttons on your phone are not working. Before too long, you rightly conclude that there's something wrong with Asterisk. Correctomundo! If you want the technical reason for why you may have lost DTMF signalling, take a look at the RFC. To put it down where the goats can get, if you go into a Chinese restaurant where only Chinese is spoken and you happen to only speak English, chances are you may leave hungry. In the world of touchtones and Asterisk, there are several different dtmfmode settings. There's one for your phone to communicate with your Asterisk server, there's another for your server to communicate with your phone, there's another for your Asterisk trunk to communicate with your provider, and there's another for your provider to talk to you. Now multiply all those combinations by two for communications with another party, and you'll have some idea of the technical hurdles... even with a perfect connection between Party A and Party B. In short, perhaps you just should be thankful you can hear the person at the other end of the call at all.

If different portions of the call are using different DTMF settings and with some compressed codecs, the touchtones cannot be deciphered at the other end of the call. There are several things you can do to improve your chances of DTMF tones working. First, use a reliable provider and buy decent phones. Second, set your server trunks, extensions, and your phones to dtmfmode=rfc2833 and see how it goes. If you still have problems, try adjusting the dtmfmode settings on just your phone and extension to some other value supported by your phone. These two must match. Try dtmfmode=inband and dtmfmode=info. Next, make certain that the dtmfmode setting for your trunk matches what your service provider is using to communicate with your server. This pair of settings must match as well. If you still don't have any luck, try a little Googling for the dtmfmode for your phone type and your provider. If it worked for someone else, chances are it will work for you. If all else fails, try another phone or a more reliable telephony service provider. Assuming you can understand them, you typically can tell whether your service provider understands the problems within about 30 seconds after the music on hold ends... which brings us to our favorite topic.

My Telephony Provider SUX. Yes. There are telephony providers and then there are telephony providers. As with most things in the world, you get what you pay for. Cheap telephony rates don't always mean crappy service, but it certainly improves your chances. All-you-can-eat plans are notoriously dangerous. Even if the telephone service is fairly good, the terms of service typically are shocking. Some even force you to agree to pay exorbitant backdated fees plus attorneys' fees if they, in their sole discretion, determine that you have used your plan for unauthorized calling.

We've got some tips that we repeat often so if you've heard them already, skip along to the next topic.

• Rule #1: If your business depends upon incoming telephone calls, don't use VoIP telephony service for all of your incoming calls. What you may want to do is order a single business line from AT&T and take Marty Tennant's advice: "You can order an arrangement called 'call forward/busy multi-path' from AT&T (confirm this beforehand) that will allow multiple call forwarding instances to another number (the VOIP one in this case)."
• Rule #2: Do some reading on which providers have good reputations. We also have a good list of providers that we regularly recommend.
• Rule #3: With pay-as-you-go termination providers for outbound calls, it doesn't cost you a dime to have numerous trunks provisioned and working on your Asterisk system. If a termination fails using your preferred provider, Asterisk will simply drop down the list until it can successfully complete the call. So don't ever put all your eggs in one basket for terminations.
• Rule #4: All-you-can eat incoming service with a free DID is still a very good deal at least in the United States and Canada. See our list for suggestions.
• Rule #5: Toll-free numbers no longer have to be expensive. See our recommendations for reasonably priced toll-free numbers, and give your business a shot in the arm for almost nothing!

What Happened to CallerID? CallerID really is the last vestige of the old Ma Bell monopoly. CallerID numbers are easily deciphered on almost all Asterisk systems regardless of your DID provider. This is true on inbound and outbound calls. CallerID name is a different story. The short answer is that the Baby Bells all maintain their own telephone directories. And chances are you're not in it if you're using VoIP telephony service. These companies seek to preserve their telephone monopoly by *NOT* processing CallerID names that are received from "foreign" systems. Instead, they take the CallerID number that is provided and look up the name in their proprietary directory. No entry = No CallerID Name display. So... the short answer is that, for outbound calls from your system, it does no good to send CallerID Name information. Almost every provider throws it in the bit bucket.

That still doesn't explain why you can't get CallerID names for incoming calls. Here's where your DID provider matters. Some of them subscribe to baby Bell-supported service that provides the names, and others don't. If your DID provider doesn't, then you can either set up your own service to supply CallerID name information, or you can get a new DID provider. For the best homegrown CallerID name service, we recommend Ultimate CNAM from Titanous. It works well on all PBX in a Flash systems and is extremely flexible in the choices provided for name lookups. It currently supports eight lookup providers: AsteriDex, WhoCalled.Us (registration required), Whitepages.com, AnyWho.com, Canada411.com, Google Phonebook, TelcoData (Ratecenter), and Fonetastic (Ratecenter).

My Passwords Don't Work Any Longer. What is it about Asterisk that makes everyone want to screw around improving their passwords? Leave them alone! So long as your initial root password is secure, you're absolutely safe from intruders except someone with physical access to your machine (even on the Internet) if you just do the following. First, using a web browser, go to the IP address of your new server. Click on Administration and then Menu Configuration and enter an Admin password that is as secure as your root password. Second, open FreePBX and click on Setup and then Administrators. Change the password for admin to something equally secure. Third, go to the Linux command prompt. Type each of the following commands and enter a secure password for each.

passwd-maint
passwd-amp
passwd-meetme
passwd-webmin

Now leave your damn passwords alone for at least six months unless you are tortured and forced to reveal all of your innermost secrets. If the annoying FreePBX password reminders bug you, then go to this link and follow the instructions to make the reminders disappear. Then leave your system alone for a week to make sure everything works reliably. Now you're free to add one new thing every other day checking often to make sure it didn't break something that was previously working. When you add ten new things at once, it's virtually impossible to put Humpty back together again. But, of course, you knew that. Enjoy!

PiaF Without Tears. Ben Sharif's PiaF Without Tears tutorial (all 208 pages) was released last week. For those that haven't yet taken a look, you're missing a treat!

Coming Attractions. With the new PBX in a Flash 1.2 release, there now are four different versions of Asterisk that can be installed: 32-bit Asterisk 1.4, 64-bit Asterisk 1.4, 32-bit Asterisk 1.6-beta, and 64-bit Asterisk 1.6-beta. Next week we'll address the installation issues with the Nerd Vittles applications using each of these new systems and expose a few more potholes in the Asterisk minefield. And we may have a new AsteriDex 4 add-on for you as well.

Nerd Vittles Cepstral Demos with Allison TTS (courtesy of les.net). You now can take some Nerd Vittles projects for a test drive... by phone! And it provides a good example of the VoIP quality you can expect with hosted service from Aretta Communications. The current demos include all five new applications preconfigured for Cepstral with the Allison TTS voice: (1) MailCall for Asterisk with password 1234 (retrieve POP3 email by phone), (2) NewsClips for Asterisk (latest news headlines in dozens of categories), (3) Weather Forecasts by U.S. Airport Code, (4) Weather Forecasts by U.S. ZIP Code, and (5) Worldwide Weather Forecasts.

The WalMart Special. We continue to believe that the Everex gPC (aka The WalMart Special) is an almost perfect server for Asterisk implementations with less than 30 simultaneous calls and up to 100 or so extensions. At $199, you can't beat the price. To make things even easier, we will have a preconfigured 2-CD ISO installation set for either the 32-bit Asterisk 1.4 or 1.6-beta version of PBX in a Flash in the next few weeks. It'll include all of the Nerd Vittles goodies plus a full system automatic backup system. All you'll need to add is a 4GB flash drive (about $15) for your weekly backups, and you'll never have to worry about losing your system again! So order your unit, and you'll be ready for the rollout. Here's the WalMart link and the NewEgg link for the latest hardware version. Add a gig of RAM for $25, and you'll have the perfect telephony server platform to begin your Asterisk adventure.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 

Some Recent Nerd Vittles Articles of Interest...

1. Join the following line as part of the line above when you see the ↩ character in the code. [↩]

Today we’re taking a Margarita Break from our shiny new PBX in a Flash 1.2 server to play with AOL’s new AIM® Call Out. AOL actually introduced the service as an Open Voice API, but it walks and quacks like a SIP termination gateway so that, of course, tempted us to try it. Since it is SIP-compatible, we thought it would be fun to see if we could get it working with Asterisk®. It didn’t take long. This is a great opportunity for all of us that live and breathe Asterisk because of AOL’s huge infrastructure. You can sign up from anywhere in the world and call over 200 countries. Plus, their pricing is competitive. Worst case: It provides another layer of redundancy for every Asterisk-based telephony system. Complete rate table is available here.

Prerequisites. If you want to follow along with our five-minute setup, then you’ll need a LAMP-based Asterisk server with FreePBX. PBX in a Flash, trixbox, and Elastix all should work. You’ll also need an AOL account if you don’t already have one. Then you’ll need to sign up for the AIM Call Out service before we reconfigure a trunk in Asterisk to support the AOL offering. Nothing in the AIM Call Out Terms of Service precludes use with Asterisk so long as it is for home or small business use and not for resale of phone service. This is not legal advice, blah, blah, blah…

Getting Started with AIM Call Out. If you don’t already have an AOL account, you can sign up for free. Or just pull out one of the thousands of CDs they mailed you and register. Once you have an AOL account name, head over to the AIM Call Out web site and register for the service. Don’t use your AOL password for your AIM Call Out service. After all, this is SIP. You will need a credit card because you’re signing up for pay-as-you-go minutes. Five bucks will get you started. The published rates are reasonable at 1.7¢ per minute for U.S. calls and 2¢ per minute to Canada, Mexico City, England, China, Germany, and Australia. We used the word "published" advisedly…

AOL Math: 1.7 + .3 = 4   AOL has taken a page from Ma Bell in terms of creative mathematics. With each call, AOL first rounds UP the time of the call to the next minute and then rounds UP the total price to the next penny. Here’s the way their Terms of Service describe it: "For point of clarity, the rounded up minutes are multiplied against the current rate effective at the end of the call (generally based on the location the call is placed) and then rounded up to whole cents (USD)." So a 70-second call in the U.S. (which should cost under 2¢ at 1.7¢ per minute using Plain Old Math) actually is billed to you at 4¢. Charitably speaking, it’s creative to advertise the cost of a call in the U.S. as 1.7¢ per minute with all the rounding that is taking place. For short calls, it can be more than double that rate once you factor in AOL’s double rounding. In our example, the 70-second call is first rounded up to 2 minutes. And then the cost of the call is computed at 3.4¢ for the already rounded up call. Then the 3.4¢ computation is rounded up to 4¢. So you see 1.7 + .3 really does equal 4 in the bowels of AOL. And welcome to a class action law suit about a year down the road if we were betting. You’d think the bean counters would learn sooner or later that you can’t publish a 1.7¢ per minute rate in the headlines and then bury a very different pricing scheme in the fine print. Ma Bell, of course, turned rounding into an art form, and AOL has simply picked up where the Baby Bells and the cell phone companies left off. And, lest we forget, six months of non-use, and your minutes (and money) disappear just like with Skype. There also are some weird rules about how much money you can spend for the first several months of use. That’s probably a good thing. They get to watch how you behave, and you get to carefully evaluate their service and performance. In the meantime, please join us in asking AOL to clean up their act and return to performing simple addition the old-fashioned way. </whining>

Now, where were we? Once your account is set up, you’ll need three pieces of information: your userid, your password, and the name of the AIM SIP gateway: sip.aol.com:5060.

Configuring an AOL Trunk in FreePBX. With relatively recent versions of FreePBX, there are two steps to get your new termination service with AIM Call Out functioning. We need to add a Trunk. And then we need to add an Outbound Route to your dialplan. To add the new trunk, click on Setup, Trunks, Add SIP Trunk in the FreePBX web interface. You can leave the General Settings, Incoming Settings, and Registration blank for now because AOL won’t let you set your CallerID and AOL won’t sell you a DID… yet. You might want to include your CallerID info just so it’ll be there if AOL ever enables the feature. For now, when you dial out through AOL, your callees will get a number displayed which, if called, plays a 30-second ad for the AIM Call Out service and then hangs up.

For the Outgoing Dial Rules, we set up our AOL service so that we dial a 6 prefix to use AOL. Then you must dial a "1″ and area code and number in the U.S. to complete a call. For other countries, you’d dial the country code, city code, and number. So our entry for U.S. calls looks like this:

6|1NXXNXXXXXX
6|1+NXXNXXXXXX


The important piece to get things working is the Incoming Settings. Name the trunk AOL or whatever you prefer. The remaining PEER Details should look something like the following using your userid and password obviously:

dtmfmode=rfc2833
host=sip.aol.com
insecure=very
nat=yes
secret=YourPasswordHere
sendrpid=yes
type=friend
username=YourAOLaccountname@aim.com


Once you save your trunk settings and reload the dialplan, then choose Setup, Outbound Routes, Add Route. Again, we’re using a dialing prefix for this service so you’ll want to be sure you move the route toward the top of your list of outbound routes when you’re finished to make sure it snags all outbound calls starting with 6 and 10 or 11-digit numbers. Here are our settings for the outbound route:

Route Name: aol
Dial Patterns:
61NXXNXXXXXX
6NXXNXXXXXX
Trunk Sequence: SIP/AOL


Save the settings and then move the outbound route to the top of your route listing. Then reload the FreePBX dialplan, and you’re done.

Overall Impressions. We’d rate the service and call quality as pretty good based on a quick look. The pricing is reasonable except on very short calls, and the Terms of Service aren’t overly offensive. In fact, if we were betting, we’d predict that AOL will offer an all-you-can-eat plan in the very near future. For now, when your call credit is getting low, you will get an email as well as a whispering voice in your ear that you’re about to run out of minutes. Could be worse. We also had no problem placing multiple, simultaneous outbound calls through the service. That’s good news for Asterisk users. For an excellent review and some good tips on AIM Call Out generally, see Dan York’s blog. Enjoy!

Late-Breaking News Flash. Ben Sharif’s PiaF Without Tears tutorial (all 208 pages) has just been released. Forget about the contents. You know that’ll be terrific like all of Ben’s previous works. But wait until you see the cover!

PBX in a Flash 1.2 Update. The response to PBX in a Flash 1.2 continues to be a bit overwhelming. If you read last week’s article, you already know that, for the first time, there now is a turnkey LAMP install that supports 32-bit and 64-bit versions of CentOS as well as the latest 32-bit and 64-bit releases of Asterisk 1.4 and Asterisk 1.6-beta. Now’s your chance to really do some experimentation. The performance of the 64-bit install is eye-popping on both a dedicated machine and running under VMware. And the Asterisk 1.6 beta has some new features that are well worth a careful look. Don’t use it in a production environment obviously!

For the first time, you now have to be careful in loading new applications because many 32-bit apps don’t play nicely (or at all) on a 64-bit version of Asterisk and CentOS. Here’s a gotcha that you need to watch out for. You can’t use a 32-bit version of Cepstral or a 32-bit Cepstral voice on a machine running the 64-bit version of PBX in a Flash. And you can’t load the Asterisk 1.4 interface to Cepstral on a machine running Asterisk 1.6-beta. So… we have reworked our tutorial on Cepstral to reflect the proper download locations for all four new flavors of PBX in a Flash 1.2.

And, of course, Digium® has broken some applications that worked just fine under Asterisk 1.4. In the next few weeks, we will provide a road map for all the Nerd Vittles applications and what it takes to get them running under Asterisk 1.6. But, if you want the short answer, everything including MailCall works on both 32-bit and 64-bit versions of Asterisk 1.4. And everything including MailCall works on 32-bit and 64-bit versions of Asterisk 1.6 once you execute the following command after installing all of the Nerd Vittles apps you wish to use. MailCall takes some special care to set up. See this message thread.

sed -i 's|\||\,|g' /etc/asterisk/extensions_custom.conf

One might ask why it was necessary to (again) ruin existing dialplans by changing vertical bars to commas. Another brilliant performance enhancement in Asterisk 1.6 courtesy of Digium and the Asterisk Development Team. NOT!

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 

Some Recent Nerd Vittles Articles of Interest…

In our last column, we lamented the fact that Asterisk® 1.6 development seemed to be on a collision course with the dinosaurs because of developer insistence on deprecating and removing commands from the application programming interface (API) in the name of technology enhancement. The problem this poses is that applications and dialplans written for previous versions of Asterisk no longer function even though the code is barely a year old. In the corporate and government sectors, this would mean major, costly (annual) rewrites of code just to keep a functioning phone system. And, as we noted, these organizations buy phone systems to last a decade so such a development strategy would all but rule out use of Asterisk in the Fortune 500, medical, and government sectors.

Today we want to share the Digium® response and address some of the new issues that have been raised. For those of you that haven't met him, Jared Smith, who co-authored the terrific Asterisk: The Future of Telephony books, now serves as Digium's Community Relations Manager. Jared sent us a thought-provoking response which you can read in its entirety here. For ease of understanding, we're going to quote a number of sections of Jared's response and address them below so that you get the full picture of how dangerous the Digium development approach is to the future of the Asterisk project. We've been concerned in the past with Fonality's decision to keep trixbox ce on the bleeding edge while reserving a more stable Asterisk product for paying customers. Now it appears Digium has decided to do much the same thing with the open source version of Asterisk. That's unfortunate for all of us that care about the future of the project.

Jared Smith: "I think we can both agree that the feature set is an important part of any PBX system. Or, as you put it, "It's the Feature Set, Stupid!" There are two major reasons for moving from Asterisk 1.4 to the upcoming Asterisk 1.6 release at all, and the first one is features. Asterisk 1.6 brings a lot of new features to the table over what was available in Asterisk 1.4 and Asterisk 1.2. (The other big change in Asterisk 1.6 is that a lot of its internal plumbing got re-worked, so that it should be more efficient, more stable, and better able to handle larger call volumes.) Unfortunately, your article doesn't differentiate between features of Asterisk and features that third-parties (yourself included) have bolted on to Asterisk. To use the same analogy that I gave when I met you in Charleston, we here at Digium want Asterisk to be the best engine in the world. Whether you make that engine into a Formula One race car or a big brown delivery truck is up to you -- we're simply building the best engine we can. Now, we've gone and built a newer version of the engine ("More horsepower! Higher torque! Faster zero-to-sixty speed!"), and suddenly everyone complains that the starter motor doesn't fit in the same place that it used to. I know it's not a perfect analogy, but hopefully you get my point... "

Uncle Ward: We obviously applaud enhancements which make Asterisk "more efficient, more stable, and better able to handle larger call volumes." It's the rest of the paragraph that highlights the fundamental problem with the current Asterisk development strategy. The point is that, for Asterisk to survive, the developers need to appreciate that they're not building a mousetrap in a vacuum. Asterisk without a dialplan is worthless. Asterisk without application code has little value particularly in vertical markets. To carry Jared's engine analogy one step further, the concern is not about Digium's repositioning the starter motor. It's about eliminating fundamental components that businesses rely upon to keep their communications engine running. It does little good to develop "the best engine in the world" if this year's version requires kerosene while last year's version ran on gasoline and next year's version requires hydrogen. Such changes force a complete reworking of the infrastructure that organizations rely upon to keep their cars and their phone systems functioning.

Jared Smith: "APIs change when major versions of the software are released. (APIs are Application Programming Interfaces -- think of them as building blocks inside of the Asterisk code that both Asterisk and third-party programs can use to do various things.) The problem is, when we make Asterisk better, we often have to change those APIs to do so... I'd challenge you to find any major project that provides source-level API compatibility as a *guarantee* between major release versions. (Look at Apache 2.0 - 2.2, PHP 4 - 5, MySQL 4 - 5, PostgreSQL 7 - 8. They all have the same thing -- Major changes almost always require API changes.) When the Asterisk APIs stop changing from major release to major release, then Asterisk *WILL* be as dead as the dinosaurs are. "

Uncle Ward: We defy anyone to explain why "making Asterisk better" required breaking every dialplan on the planet because some developer thought Set(TIMEOUT(digit)=timeout) was a code improvement in Asterisk 1.4 over DigitTimeout(timeout). No one wants to stand in the way of progress. But moving forward is quite different than throwing the baby out with the bath water. Supporting both syntaxes would have required one extra line of code in the API. In the alternative, Digium could have released a source code translation application which would automatically convert existing code to the new syntax. This almost always has been done with major changes in programming languages. We would hasten to add that most of the developer-inspired changes with which we have been concerned have little or nothing to do with making Asterisk a "better engine." It's just a different engine. And therein lies the problem!

Jared Smith: "Luckily, Asterisk is an open-source project, which means that when Asterisk does evolve, that the changes aren't made in secret. Any third-party developer who wants to make sure his code remains compatible with the latest version of Asterisk can do so at any time. He doesn't have to wait until 1.6.0 is released to find out that his code will have to be changed to fit the new APIs. The Asterisk code is always available to test, play with, qualify against, etc. so that the developer can update their code to be compatible, so that when the time comes that real users want to use it, their applications will be ready."

Uncle Ward: The concern here isn't that third-party developers can't make changes to accommodate future Asterisk API changes. The problem is that businesses that stake their livelihood on a phone system that is Asterisk-based expect it to keep working year after year after year. Third-party developers come and go. So, if a company purchases an Asterisk-based system which includes fax and text-to-speech telephony support, those companies have a right to expect that their applications will work next year with the currently supported version of Asterisk. Jared's response sent me looking for the image to accompany this week's article: an ostrich burying his head in the sand. Third party developers move on or die, Jared. You can't pretend that folks never used their code because you're too focused on future enhancements to your race car engine to worry about preserving the necessary infrastructure to support applications that already work. As we put it last week, "You break it, you fix it. I break it, I fix it." That's a really simple design concept that should be fundamental to any API development changes. This in no way impedes the design goal of "making things better." Just don't make other things worse in the process.

Jared Smith: "The next point I'd like to address is that of responsibility. Your article somehow assumes that it's the responsibility of the Asterisk developers to somehow know about all these third-party apps, and make sure they never break due to API changes. I can see three flaws with that argument -- first of all, there's no way the Asterisk developers could possibly know of every third-party application, it's state of affairs, and so forth... The second flaw is this -- even assuming for a moment that we could keep track of all the third-party apps and try to keep them up to date (which we both know isn't possible), licensing concerns would keep the Digium-paid Asterisk developers from doing so... The third flaw to that argument is the point I made earlier... if Asterisk *were* to guarantee source-code API compatibility between major releases, there's no way possible that Asterisk could continue to grow, evolve, and adapt to the changing telephony market. "

Uncle Ward: I'm reminded of the Venus and Mars book about the differences in perspective between men and women. Are you really saying that Asterisk developers had no idea that folks were using dialplans and text-to-speech applications with Asterisk after Digium just worked with Cepstral to produce an Allison voice?? Come on, Jared. This isn't about whether it is Digium's responsibility to fix third-party developer code. This is about whether corporations and government organizations are going to invest in a telephony system when the business philosophy of the engine manufacturer is that they could care less whether they break existing telephony applications with each new product release. As I read Jared's response, Asterisk developers can't and won't be responsible for making sure they don't break existing applications and dialplan code, and Digium won't do anything to migrate existing code to new platforms. I'm not sure I understand how development of a piece of migration application code requires a knowledge of every third-party application in the universe. Presumably, the Asterisk development team does know when it changes the syntax of some command in the existing API. Why then would it be so difficult to provide another application that translated the "old code" into the "new syntax?" That doesn't require that any third-party apps be reviewed. And it doesn't stymie future development. Just provide the tool to fix stuff that you broke!

Jared Smith: "Last but not least, let's talk directly about your bug report. In it, you claim that 'Lack of native support for either Flite or Cepstral TTS breaks thousands of existing text-to-speech Asterisk applications.' Asterisk has never had native support for either Cepstral or Flite for text-to-speech, so I'm not sure how not having it in Asterisk 1.6 breaks anything. I'm afraid that if I were to follow your logic to its logical conclusion, it would be better to write that as 'Since the developer that wrote app_swift won't update the code for Asterisk 1.6, it's Digium's responsibility to do so.' Again, I've got to point out that Flite and app_swift are totally outside the control of the Asterisk development team."

Uncle Ward: Wrong again. What the Asterisk developers can control is making sure that, when a change is made to the API, they either support the new and old syntax or, in the alternative, that the developers provide a separate tool to convert existing source code to the newly-created syntax. That's what any responsible developer would do. This isn't a problem with a third-party application developer refusing to update his or her code. Many of these developers are no longer available or reachable. So it behooves the proponent of changes that impact the operation of existing telephony systems to provide a migration vehicle to the new platform. It's as simple as that. Give it some more thought, Jared. There's a lot more than either of us may appreciate riding on the outcome of our discussion.

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 

Some Recent Nerd Vittles Articles of Interest...

We’ve patiently waited until after April Fool’s Day to publish this column, but we’re having second thoughts. It may have been more fitting yesterday. One of the problems with laying track in front of a steaming locomotive is that someone still needs to watch where the train is headed. So it is with Asterisk®. And 1.6 has all the ingredients of a train wreck waiting to happen. To fully appreciate the reality of the situation, one need look no further than the business model of the Ciscos, Avayas, and the Nortels. Simply put, no customer cares what version of a phone system they are buying. Or, to dumb it down to a Clintonism: "It’s the Feature Set, Stupid!" When the features stop working, the customers start walking. It’s as simple as that.

When we began the PBX in a Flash project last November, our emphasis was radically different than some of the other Asterisk aggregations. First and foremost, we wanted a product that was stable. Of equal importance was our own Big Easy: easy to use, easy to enhance, and easy to upgrade. We didn’t want users or VARs having to reinvent the wheel each time a security patch or new enhancement was released. 40,000 downloads in just over four months tells me we got it just about right. To look at it from the customer side, no business (that wants to stay in business) will tolerate a phone system that is routinely out of service for upgrades much less one that takes away features that the business depends upon. Whether it’s Caller ID, or Text-to-Speech, or Screen Pops, or Conferencing, or Phone Blasting, or even a Call Center really doesn’t matter. It does no good to tell a customer that they lost critical functionality but now they have the latest version of X. You can add your own customer expletive here if you’ve ever tried this approach in the real world.

Which brings us back to Asterisk 1.6. In the good old days when there wasn’t much of a feature set and when no business would stake their livelihood on Asterisk, it really didn’t much matter when a new version of Asterisk was released. To put it charitably, things could only get better. Well, things have changed. Businesses now rely upon Asterisk. So the dynamics are quite different. It’s no longer acceptable to trash big chunks of code without making certain that you didn’t break something that was already working. It’s no longer acceptable to invent new verbs in the programming language while deleting commands that used to work. We defy you to find a link to any document that explains the transition from Asterisk 1.2 verbs to Asterisk 1.4 produced by the developers of the product. Asterisk 1.6 continues the programming carnage while adding some bells and whistles of its own: for example, an entirely new and different Asterisk Manager. And the scorecard: Screen Pops, Dead. Phone Blasting, Dead. Flite Text-to-Speech, Dead. Cepstral, Dead. Speech-to-Text, Dead. To show you the mentality of the programmers that think all of this is a good idea, here’s the response to our complaint that Asterisk 1.6 broke virtually all existing text-to-speech applications… again!

Summary: 0012348: Neither Flite nor Cepstral TTS works with Asterisk 1.6

Description: Lack of native support for either Flite or Cepstral TTS breaks thousands of existing text-to-speech Asterisk applications.

Response: This is clearly code that is not in Asterisk. Many of us cannot even look at the code, unless it has been disclaimed. If the original developers are not willing/able to update their code, then you are going to either have to find somebody who will do it for free, or offer a bounty for somebody to do it. This is most certainly not the place to be requesting this. In the future, before posting any bug reports, please read the bug guidelines as linked on the main page of bugs.digium.com.

Wrong, wrong, wrong. That’s the type of attitude that will sound the death knell for Asterisk. Here’s the tattoo that should be stamped on every programmer’s foreskin forehead: You Break It, You Fix It. I Break It, I Fix It. Hopefully Mark & Co. will come to their senses before it’s too late.

Click here for Chapter 2.

Footnote: Since releasing this article earlier today, we’ve gotten a response from Cepstral Support. They also had contacted Digium® for help with this. If you loved the original Digium response to the bug report, you’ll really enjoy this one:

"Thank you for your interest in Cepstral Voices. In my discussions with Digium they made three comments:

1) That releasing an "ISO" of Asterisk may break the GPL2 (they were more certain than "may"). I would check with Digium on this. (WM: We already have sent the correspondence and can’t wait to hear more!)

2) Asterisk V. 1.6 is in Beta and that they take the typical corporate stand on Beta. They know of incompatibilities, but since it is in Beta – they are working through these.

3) Cepstral and Digium both recommend that you contact the people that wrote the app_swift layer. In the future there may be some app_swift / app_cepstral / res_cepstral that is "official" – but right now it is a bit pot-luck in support."

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

BOGO Bonaza: Enjoy state-of-the-art VoIP service with a $10 credit and half-price SIP service on up to $500 of Skyetel trunking with free number porting when you fund your Skyetel account. No limits on number of simultaneous calls. Quadruple data center redundancy. $25 monthly minimum spend required. Tutorial and sign up details are here.

The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 

Some Recent Nerd Vittles Articles of Interest…

Today we're pleased to introduce TeleYapper 4.0, an updated, Asterisk® 1.4-compatible version of our telephone broadcasting service (aka phone blasting software). For those with text-to-speech capabilities on your PBX in the form of either Flite or Cepstral, today's addition adds support for individualized, text-based messages to everyone in a group distribution list. And, for those with multiple outbound trunks, TeleYapper 4.0 supports simultaneous calls using multiple trunks. Version 4 works with PBX in a Flash, of course. And it should work well with trixbox 2.x and later versions of Asterisk@Home. If you're using an earlier version of Asterisk@Home, see our previous versions and tutorial.

For those that have never used TeleYapper, let us give you a quick summary of the product. It's an automated message broadcasting service commonly known as a call blasting or phone blasting system. In addition to loads of creepy uses, phone blasting has a legitimate purpose. TeleYapper is licensed for the following uses: to send prerecorded phone messages for neighborhood association announcements, school closings, tornado alerts, little league practices, fund raisers, municipal government reminders, and for just about any other non-commercial purpose. We'll have more to say about the licensing restrictions on this product in a minute.

Everything you'll need to get TeleYapper 4.0 dialing away is in this article. And functionally, TeleYapper still works identically to prior versions with a few embellishments. For those new to TeleYapper, here's how. You create a recorded message using Asterisk. Then you create a list of phone numbers to call in a MySQL database using a tool such as phpMyAdmin which is bundled with PBX in a Flash and some other Asterisk distributions. Once your distribution list is set up, you place a phone call either to kick off TeleYapper or to redial calls that failed the first time around. The software will dutifully swing into action and call qualifying phone numbers from any of ten calling categories that you specify when you set up your database of message recipients. TeleYapper then will deliver the message you've recorded. It works much like call-em-all.com and numerous other telephone broadcasting services with one important difference: TeleYapper is FREE! So, instead of paying 15¢ a call or $35 to $100 a month for a commercial service or spending thousands of dollars for a commercial dialer, now you can do it yourself using TeleYapper and your (also free) PBX in a Flash system. Look at the top of this page (just below the Nerd Vittles header) for links to Windows, Mac, and Linux versions of PBX in a Flash that are perfectly suited for use with TeleYapper. Today we'll actually get TeleYapper making calls and emailing you the results of those calls. Don't be intimidated by this technology. You can complete this project in under 5 minutes. If you're using one of our PBX in a Flash systems, then installation of TeleYapper 4.0 is a one-minute install. Either way we've done most of the hard work for you. All you'll need to do is create your call distribution groups.

Legalese. For those that are used to buying flawless software such as Microsoft Windows or Microsoft Office, let's be sure we're all on the same page up front. First, you're not buying this software. It's FREE! And, yes, sometimes you get what you pay for. Second, don't assume today's version is error-free. It's probably not. But we try pretty hard to write reliable code. Third, by downloading or using this software, you are agreeing to assume all risks associated with use of the software. NO WARRANTIES EXPRESS OR IMPLIED INCLUDING ITS FITNESS FOR USE OR MERCHANTABILITY ARE PROVIDED WITH THIS SOFTWARE. And, finally, read or reread our previous article concerning Do Not Call statutes in your jurisdiction. Make sure you are in compliance before placing any calls. Failure to heed this advice may subject you to serious criminal and civil penalties. If any of this gives you heartburn, exercise your constitutional right to not use the software.

Overview. TeleYapper 4.0 also provides a good framework for anyone wanting to write Asterisk AGI scripts using PHP. The code is well-documented to demonstrate how to pass variables to an AGI script from your dialplan and how to retrieve variables from an Asterisk AGI script into your dialplan. We needed this for TeleYapper because we're using a phone call to an Interactive Voice Response (IVR) session embedded in the dialplan to begin the calling process. We use the IVR session not only to determine which group of callees to call but also to give the caller the option of placing a call to everyone in the group or just those to whom the initial call was unsuccessful. After the caller hangs up, the results are passed to the teleyapper.php application to do the heavy lifting. The PHP program takes advantage of an AGI script's ability to actually set dialplan code in motion once a call is answered. In order to log calls and track which ones are successful, we have to pass variables into that dialplan code and then execute another PHP script when the call is completed. Stated another way, every call requires two round-trips from the Asterisk dialplan to PHP/AGI scripts. So, if you can't figure out how to pass variables back and forth using this application, you probably should consider another line of work. For those that just want to use the TeleYapper application and not learn much of anything about programming, you're welcome to do that subject to the license agreement which follows. We hope you'll put it to good use for the betterment of a school, an intramural sports program, or a neighborhood or community in which you happen to live.

Licensing. We are retaining ownership of this software as well as the copyright. It is licensed for use under the terms of the Creative Commons Attribution Non-Commercial license. A Plain English summary is available here. We've done this primarily to do our part to stamp out the telemarketing creeps of the world. Those wishing to use TeleYapper in a commercial environment must first request and purchase a license outlining your proposed terms of use. We will promptly respond with a yay or nay. Telemarketers need not apply!

TeleYapper in a Nutshell. Before we get to the code, let's briefly cover how this message broadcasting system works. When you dial M-S-G (extension 674) from a phone connected to your PBX, TeleYapper will answer and prompt you for your password. Once you correctly enter the password, an interactive voice response (IVR) system will swing into action and give you several choices. That's what the [yapper] context handles. Pressing 1 lets you listen to your prerecorded TeleYapper message (if you have one). You don't yet so don't press 1. Pressing 2 lets you record a new TeleYapper message. This is handled by the [yapper2] context. Do this first and record something ... anything. You can rerecord a new message at any time by choosing option 2 again. Pressing 3 lets you kick off a TeleYapper dialing spree. It's handled by the [yapper3] context. Don't do this until we add and you populate your new database below, or you'll get smoke. When you choose option 3 to initiate a TeleYapper calling session, the system will first prompt you for a group option number to use. This is managed by the [yapper-options] context. Simply stated, when you build your database of callees for TeleYapper, you can specify a one-digit group number for each entry in the file. Then, when you begin a calling session, you can narrow down the calling group by telling TeleYapper which group of callees to call. If you want a callee to be in more than one group, you simply enter that callee into the database multiple times with different group numbers. If you want everyone in the same group, then enter 0 for every person in your database.

Once you specify the group number during your TeleYapper session, the system will actually look up and report back how many messages will be delivered to the callee group you've chosen. Allison will say something like this assuming there were 146 calls to be placed: "The number I have is one hundred and forty six messages." This will give you the count of qualifying records in the database and the option of proceeding with the calls, canceling the transaction, or just redialing the numbers of the calls that failed to this group on the previous pass through the database.

TeleYapper's Calling Process. For those that like lists, it may help to visualize how all the TeleYapper code fits together by laying out the actual program steps in a typical call:

• Caller with TeleYapper password places call to M-S-G (extension 674) to activate a TeleYapper session.
• Asterisk answers the call, provides IVR menu choices: playback a message, record a new message, or place a call.
• If caller chooses to place a call, IVR prompts for Group number to call (0-9).
• Asterisk passes the Group number to MySQL (checkgroup.php) to look up the number of callees in the chosen Group.
• Group count is passed back to Asterisk which uses Allison to tell the caller how many callees are in the chosen Group.
• Caller has option of placing the call, hanging up, or choosing to redial previously unsuccessful calls in the chosen Group again.
• If caller chooses to place a new call, Asterisk thanks the caller, hangs up, and then passes control to teleyapper.php to handle placing the calls.
• TeleYapper time stamps dialing scripts two minutes apart for each call beginning two minutes after the initiating request. Scripts are placed in the Asterisk outgoing calls queue.
• TeleYapper initializes the date/time and status fields for each record in the Group to be called. These are only filled in when a call is then answered.
• If you've enabled logging in teleyapper.php, then the log is generated after all of the call setups have been completed.
• If you've enabled emailing of the teleyapper.php log, then the log is emailed to your email address at the same time.
• Asterisk checks its call queue each minute and places each call at the appointed time. Then it waits for the callee to answer.
• If no one answers the call, nothing is posted to the MySQL database regarding call completion. That's how we identify unsuccessful calls.
• If the call is answered, the callee is advised to hold for an important message. If the msg field for this callee is blank, then the prerecorded message is played to the caller.
• If the msg field has a message in it, then either Flite or Cepstral is used to read the custom message to the callee.
• Callee is then prompted to press 1 to acknowledge the call and hang up, press 2 to replay the message, or press 3 to remove the callee from the database.
• If callee presses 1, Allison says goodbye and Asterisk hangs up the call. MySQL database will show date/time of call with status of OK.
• If callee presses 2, your message is replayed, and then the call is disconnected. MySQL database will show date/time of call with ReplayedMsg as status.
• If callee presses 3, log will reflect that caller requested blacklisting. MySQL will actually DELETE this person from your database. It's the LAW!
• If callee makes no choice, Asterisk will replay your message, then hang up, and record the date/time of call with status of AnsMachine.
• If you've enabled logging in teleyapper2.php, then the individual call log is generated and appended to the main log file after each call has been placed.
• If you've enabled emailing in teleyapper2.php, then the call log is emailed to your email address after EACH call has been placed.

The TeleYapper code not only handles the actual dialing of the callees you've entered in your MySQL database (teleyapper.php), it also plays your message when a callee answers (dialplan contexts), and documents what happened during the calls (teleyapper2.php). Call progress is documented in two ways. First, when a call is completed, TeleYapper will log the date and time of the call as well as a best guess of what happened during the call in your MySQL database. So browsing entries in your TeleYapper database will always show the date, time, and status of the last completed call to each callee. We'll build a web interface for this some day.

When you install the TeleYapper PHP components, there are some configuration options which will also let you create a detailed log of what happened during the TeleYapper calls. If you have email working reliably on your Asterisk system, you also can enter your email address and tell TeleYapper to email you every log that is produced. There are log entries for the initial call setup (handled by teleyapper.php) and for the placement of the individual calls (handled by teleyapper2.php). Finally, you have the option of creating a new log with each series of calls that are placed (the default setting), or you can configure TeleYapper to keep adding to the end of the initial log. In the latter case, it's up to you to erase the log before it fills up your disk. Individual call entries, if logged, will be appended to the main TeleYapper call setup log (/var/log/asterisk/teleyapper.txt).

Installing TeleYapper 4.0 The real beauty of PBX in a Flash as an Asterisk platform is demonstrated by the ease with which you can install new applications such as this one. The drill is very simple. You download an install script, make it executable, and run it. Less than a minute later, the TeleYapper install is done. For those that want to take advantage of the new text-to-speech option in TeleYapper to deliver customized messages to callees, we've included an option in the installer that preconfigures TeleYapper for either Flite or Cepstral so you don't have to touch your dialplan. This can be changed later by replacing Flite with Swift or vice versa in the [broadcast] and [broadcast2] contexts in your extensions_custom.conf dialplan code. Here are the commands to execute to install TeleYapper after logging into your PBX in a Flash system as root:

cd /root
wget http://bestof.nerdvittles.com/applications/teleyapper4/teleyapper.pbx
chmod +x teleyapper.pbx
./teleyapper.pbx


Adding Entries to the TeleYapper Database. We use the MySQL database management system to manage the list of callees for TeleYapper to dial. It can handle a database of almost any size and generally stands up well in performance comparisons with Oracle. So you're covered on the database front.

The install script created the MySQL database to support TeleYapper. The easiest way to work with MySQL databases is to use the phpMyAdmin which is accessible through the Tools tab in FreePBX on PBX in a Flash systems. You'll need to login as maint with your maint password to access phpMyAdmin. After phpMyAdmin loads, click on the teleyapper database in the left column. Then click the teleyapper.callees table entry in the left column to open the file. Now click the Insert tab at the top of the right column to add entries to the table. You only need to add information for the name, phonenum, and group fields in the corresponding values column. The id, lastokcall, and lastcall fields should be left as is. The id field gets calculated automatically. The lastokcall will record the time and date of the last successful call using TeleYapper. And the lastcall field identifies what happened during the last call to this person, e.g. ok means the call was completed successfully, no answer means no one answered the call, or answering machine means an answering machine took the call. Only fill in the msg field if you want to deliver a customized message (rather than your prerecorded message) to this callee. And, remember, this functionality only works if you have Flite or Cepstral installed and working on your system. Flite works by default on all PBX in a Flash systems. Cepstral you have to pay for. See our Cepstral article if you'd prefer to use Cepstral with Allison's voice for this and other Nerd Vittles text-to-speech applications.

Where were we? You can add up to two MySQL records at a time with phpMyAdmin and, by clicking the Insert Another New Row button, you will be returned to this data entry screen after you save your entries by clicking the Go button. The name field allows you to quickly review entries you've made. It won't be used when making TeleYapper calls. The phonenum field is the important one. This is the exact dial string required to place a call on your Asterisk system to this callee using whatever VoIP or PSTN outbound trunk you plan to use with TeleYapper. For example, if your preferred provider requires 11-digit phone numbers with a 1, area code, and number, then that's the way the numbers should be entered into the TeleYapper database. The group field has already been discussed. Just enter a number between 0 and 9 to identify the group with whom this individual should be associated. Finally, after adding records to the table, you can click the Browse tab to review your entries. And, while Browsing, you can click the Pencil icon beside any record entry to edit it. Clicking the red X icon beside a record entry deletes the record. If, for some reason, you wish to delete ALL the records in the file, click the Empty tab at the top of the right column. Under no circumstances should you click on the Drop tab as this removes not only the table's contents but also the table structure itself. In short, you'd have to recreate the database table again.

Answering the Incoming Call. A simple addition to your dialplan is used to force Asterisk to answer calls to M-S-G (extension 674) and pass them to the TeleYapper contexts for processing. The following code has been inserted into the [from-internal-custom] context near the top of extensions_custom.conf. Be sure to change the 1234 password below to something secure for your system since this will be used to gain access to your TeleYapper system!

exten => 674,1,Answer ; dial MSG on any extension to manage your TeleYapper system
exten => 674,2,Wait(1)
exten => 674,3,Authenticate(1234)
exten => 674,4,Goto(yapper,s,1)


Once you change the password, save your changes and reload the Asterisk dialplan.

If you're using FreePBX, we recommend you also make the following addition to your FreePBX configuration. Log into FreePBX with a web browser. Then choose Setup, Misc Destination. Add a new entry for TeleYapper with 674 as the Dial entry. Save your change and reload the dialplan when prompted to do so.

Configuring checkgroup.php AGI Script. The checkgroup.php script was installed in your /var/lib/asterisk/agi-bin directory as part of the installation. This script includes a debug log. The default settings are to create a new log file (/var/log/asterisk/telecheck.txt) each time the script is executed. This doesn't take up much room and is always there for you to read if something comes unglued: cat /var/log/asterisk/telecheck.txt. There are some other options. You can turn off the log file entirely ($debug=0). You can choose not to erase the previous log file each time the script is run ($newlogeachdebug=0) in which case the file continues to grow until your hard disk fills up. And you can have the log file emailed to you each time the script is executed ($emaildebuglog=1) by adding your email address ($email=youremailaddress). The last option obviously assumes you have followed our previous tutorial and gotten outbound email working reliably on your system. The functions are controlled by the following lines at the top of the checkgroup.php file. 1 means yes, and 0 means no. Just edit the file carefully: nano -w checkgroup.php. And save your changes when you're finished: Ctrl-X, Y, then press Enter.

$debug = 1;
$newlogeachdebug = 1;
$emaildebuglog = 0;
$email = "yourname@yourdomain" ;


Configuring teleyapper.php AGI Script. The teleyapper.php script has a number of configuration options including a debug log. Edit the file carefully while positioned in the /var/lib/asterisk/agi-bin directory: nano -w teleyapper.php. And save your changes when you're finished: Ctrl-X, Y, then press Enter. All of the options are shown below.

$maxlines=1 ;
$maxretries=1 ;
$retrytime=60 ;
$waittime=60 ;
$callspread=2 ;
$debug = 1;
$newlogeachdebug = 1;
$emaildebuglog = 0;
$email = "yourname@yourdomain" ;
$trunk = "local" ;
$callerid = chr(34) . "TeleYapper" . chr(34) . " <6781234567>" ;

The first line lets you set the number of simultaneous calls which can be placed. Be sure you have sufficient outbound trunks to support the number you insert and be sure to use the "local" setting for $trunk unless you know what you're doing. The callspread variable determines the spacing of calls (or groups of calls if you have multiple outbound trunks) to your various callees. The default has been changed from one minute to two minutes based upon extensive testing with large numbers of calls. This means the call to the second callee (or group of callees if you have specified $maxlines > 1) begins two minutes after the first call starts. Because a broadcast message is usually more than about 20 seconds long, we have found the 2 minute setting to be better since it allows sufficient time to complete the first call before the next one begins. Otherwise, calls will start failing if you only have a single outbound trunk. If you're going to be placing hundreds of calls, be sure to read our previous article which covers a real-world example using a 700-call database.

The debug flags in this file are set the same way as in the checkgroup.php script above: 1 means yes, and 0 means no. The default settings are to create a new log file (/var/log/asterisk/teleyapper.txt) each time the script is executed. This doesn't take up much room and is always there for you to read if something comes unglued: cat /var/log/asterisk/teleyapper.txt. There are some other options. You can turn off the log file entirely ($debug=0). You can choose not to erase the previous log file each time the script is run ($newlogeachdebug=0) in which case the file continues to grow until your hard disk fills up. And you can have the log file emailed to you each time the script is executed ($emaildebuglog=1) by also adding your email address ($email=youremailaddress).

Two settings you will need to review and perhaps adjust to get calls to complete properly are the trunk and callerid variables. If you wish to use a specific trunk in your dialplan for outbound calls, the syntax for the outbound trunk is the same as it is in your dialplan, e.g. sip/telasip-gw or iax2/voxee. Look at the OUT settings in your /etc/asterisk/extensions_additional.conf file if you're not sure. At the request of a number of users, we've now added a new option which allows all outbound TeleYapper calls to be placed using the default dialplan rules on your server. The advantage of this approach is that different VoIP providers can be used automatically for different types of calls in your TeleYapper database. To use your default dial rules, set the trunk in all lowercase letters to local and TeleYapper will handle the rest of the setup for you.

The callerid variable should be set to the callerid number of your outbound trunk unless your service provider allows callerid spoofing (some don't!). Don't delete the variable! Just leave the default value.

Finally keep in mind that the format of the numbers to be dialed in your database must exactly match the syntax your trunk provider is expecting to see unless you're using your default dialplan rules. Otherwise, all of the outbound calls will fail. For example, if your provider requires that calls begin with a 1 followed by a 3-digit area code and 7-digit number, then that's the way the numbers must be entered in your TeleYapper database. Do NOT use hyphens or other punctuation in the phone number entries!

Configuring teleyapper2.php AGI Script. The only configuration options in the teleyapper2.php script are for the debug log on individual calls that are placed. We recommend you leave the existing settings, or you'll get a new email every time each individual call is placed by TeleYapper. You can edit the file while positioned in the correct directory: nano -w teleyapper2.php. And save your changes when you're finished: Ctrl-X, Y, then press Enter. All of the options are shown below.

$debug = 1;
$emaildebuglog = 0;
$email = "yourname@yourdomain" ;

The debug flags in this file are set the same way as in the teleyapper.php script above: 1 means yes, and 0 means no. The default settings are to append individual call information onto the teleyapper.txt log file (/var/log/asterisk/teleyapper.txt) each time a new call is placed. Unless you're planning to call hundreds of thousands of people, this doesn't take up much room and is there for you when something comes unglued. The other options are as follows. You can turn off the individual call logging entirely ($debug=0). And you can have the entire teleyapper.txt log file emailed to you each time a call is placed ($emaildebuglog=1) by also adding your email address ($email=youremailaddress). For your initial test calls, this may be desirable just so you can see what's going on ... if you're too lazy to read the log.

Changing the Initial Greeting. When TeleYapper places calls, it initially announces the call before playing your prerecorded message or customized text-to-speech messages. The default greeting is "Hi. Please hold a moment for an important message." This may not be appropriate for what you plan to do so we've added several more from which you can choose. The greetings all are stored in /var/lib/asterisk/sounds/custom. The one that actually plays is nv-yapintro.wav. So just choose the one you like and copy it over as nv-yapintro.wav. The choices include the following:

• nv-yapintro-reminder.wav - Please hold a moment for an important reminder.
• nv-yapintro-pubsvcmessage.wav - Please hold a moment for an important public service message.
• nv-yapintro-safety.wav - Please hold a moment for an important safety alert.
• nv-yapintro-school.wav - Please hold a moment for an important message from your school.
• nv-yapintro-team.wav - Please hold a moment for an important message from your team.
• nv-yapintro-message.wav - Please hold a moment for an important message.

Taking TeleYapper for a Spin. Once you restart Asterisk (amportal restart), you should have a TeleYapper System that works. First, start up the Asterisk Command Line Interface (CLI) by typing asterisk -rvvvvv from the command prompt on your system after you've logged in as root. The CLI now will track the progress of your TeleYapper sessions.

Using phpMyAdmin, add your cellphone number to your TeleYapper database and specify Group 0 for the entry. Now dial 674 and provide your password, record a message (Option #2), and then place a call (Option #3) to Group 0. Press 1 to kick off the TeleYapper calling spree. Check your CLI and TeleYapper logs if your cellphone doesn't ring in about two minutes.

Real-World Test of TeleYapper. Be sure to check out our previous article on TeleYapper for a real-world example dialing 700+ neighbors with information about a rezoning meeting.

From Our Legal Department, moi: The TeleYapper product name (our feeble attempt at humor through parody) has absolutely no affiliation with TeleZapper, the terrific hardware product designed to keep telemarketers from bugging the hell out of you while you're eating your dinner. We confess that our sense of humor got the better of us in coming up with the name for this non-commercial (aka "free") utility designed primarily as an educational vehicle to assist the Asterisk community in recognizing the almost limitless potential of AGI and PHP programming. Our parody seeks to amuse, not to confuse. Our telephony software Yaps. Their telephony hardware Zaps. Other than a telephone line, there is no product similarity as the two conjoined words make clear. And, yes, that is the whole point! The products are opposites, not identical nor even similar. One letter makes all the difference in Night and Light. So it is with Yapper and Zapper. Brand confusion in trademark law arises from synonyms, not antonyms. It is systems like what we're writing about today that TeleZapper is designed to protect against. And it does that very well. In fact, we use TeleZapper hardware in our own home and have for many years. The only problem, of course, is when that tornado comes rolling down the neighbor's street, it would have been nice to get the automated phone call from TeleYapper at the neighborhood headquarters. But, who cares, right? It's only your house. Class dismissed.

64-bit PBX in a Flash IS Ready for Testing. For those of you with 64-bit systems, you really need to try our new 64-bit version of PBX in a Flash. Some are reporting more than a 200% performance improvement over the 32-bit version. If you'd like to try out the 64-bit release, contact Joe Roper through the PBX in a Flash Web Site for a link to the download. The following processors reportedly support 64-bit code:

Intel NetBurst
Intel Xeon (some models since "Nocona")
Intel Celeron D (some models since "Prescott")
Intel Pentium 4 (some models since "Prescott")
Intel Pentium D
Intel Pentium Extreme Edition
Intel Core microarchitecture
Intel Xeon (all models since "Woodcrest")
Intel Core 2 (Including Mobile processors since "Merom")
Intel Pentium Dual Core (E2140, E2160, E2180, T2310, T2330 and T2370)
Intel Celeron (Celeron 4x0; Celeron M 5xx)
AMD Athlon 64
AMD Athlon 64 X2
AMD Athlon 64 FX
AMD Opteron
AMD Turion 64
AMD Turion 64 X2
AMD Sempron ("Palermo" E6 stepping and all "Manila" models)
AMD Phenom


Coming Soon for Bleeding Edgers. If you really like living on the edge, there will be an Asterisk 1.6 version of PBX in a Flash shortly for experimentation purposes only. I spoke with Tom King by phone using a functioning 1.6 build today, and everything worked swimmingly including most of the FreePBX 2.4 functionality. The design is the same as the current PBX in a Flash system with the latest Asterisk 1.6 beta code, of course, and all the goodies needed to support it in the familiar LAMP environment consisting of Linux, Apache, MySQL, and PHP. Stay close to the PBX in a Flash Forum for the 1.6 ISO announcement!

Do You Get It Yet? With the roll-out of these new ISOs, even the slow-learners should be starting to see why a carefully designed, source-based LAMP install of Asterisk is so compelling. Until now, not a single PBX in a Flash user ever has had to reinstall PBX in a Flash from scratch. A few button clicks is all it takes to transform the original PBX in a Flash ISO into a current, virtually bug-free system with the latest and greatest versions of Asterisk, FreePBX, and CentOS. And issues raised on our forums are addressed in minutes, not weeks or months. Pretty neat if we do say so. Give PBX in a Flash a try. You'll be glad you switched. 40,000+ downloads in four months and four days kinda says it all.

Munin Comes to PBX in a Flash. With special thanks to Andrew Gillis for his pioneering work in porting munin to the Asterisk LAMP environment, we are pleased to announce that a one-minute install of munin is now available for PBX in a Flash. For those unfamiliar with munin, it is a fantasic, web-based server monitoring and management tool that now has customized add-ons to support Asterisk.everything. You can read up on munin and download the installation script from the PBX in a Flash Forum.

Aastra 2.2 Firmware Update. For those heeding our advice and using the World's Best Asterisk Phone, the Aastra 57i, there's a terrific firmware update that's 100% compatible with the existing Nerd Vittles Aastra phone design. This update adds dozens of new features and fixes the one concern of losing the transfer and forward buttons when a second incoming call arrives. The new features include:

• Group Paging via RTP Multicast
• Call Forward and DND for each account
• One-touch Call Transfer to BLF Extensions and SpeedDials
• Intercom and AutoAnswer Enhancements
• and many, many more...

Just download the update from here, copy the unzipped 57i.st file into /tftpboot on your PBX in a Flash server, and restart your phones. It's as simple as that.

Conversational Linux for Windows Users. Last, but not least, Joe Roper, one of the key players on the PBX in a Flash Development Team, has a great Linux for Newbies tutorial that should be available in the next couple weeks. If you're a Windows user and want to learn all of the Linux basics in a simple, Plain-English tutorial, this is the document for you. The document will be made available on pbxinaflash.com and pbxinaflash.net at no cost although Joe does accept consultancy work when you have a project that demands a bit more expertise. A portion of that income is generously plowed back into the PBX in a Flash development project. Stay tuned!

Special Thanks to Our Generous Sponsors

FULL DISCLOSURE: ClearlyIP, Skyetel, Vitelity, DigitalOcean, Vultr, VoIP.ms, 3CX, Sangoma, TelecomsXchange and VitalPBX have provided financial support to Nerd Vittles and our open source projects through advertising, referral revenue, and/or merchandise. As an Amazon Associate and Best Buy Affiliate, we also earn from qualifying purchases. We’ve chosen these providers not the other way around. Our decisions are based upon their corporate reputation and the quality of their offerings and pricing. Our recommendations regarding technology are reached without regard to financial compensation except in situations in which comparable products at comparable pricing are available from multiple sources. In this limited case, we support our sponsors because our sponsors support us.

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The lynchpin of Incredible PBX 2020 and beyond is ClearlyIP components which bring management of FreePBX modules and SIP phone integration to a level never before available with any other Asterisk distribution. And now you can configure and reconfigure your new Incredible PBX phones from the convenience of the Incredible PBX GUI.

VitalPBX is perhaps the fastest-growing PBX offering based upon Asterisk with an installed presence in more than 100 countries worldwide. VitalPBX has generously provided a customized White Label version of Incredible PBX tailored for use with all Incredible PBX and VitalPBX custom applications. Follow this link for a free test drive!
 

Special Thanks to Vitelity. Vitelity is now Voyant Communications and has halted new registrations for the time being. Our special thanks to Vitelity for their unwavering financial support over many years and to the many Nerd Vittles readers who continue to enjoy the benefits of their service offerings. We will keep everyone posted on further developments.
 

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