Nerd Vittles

If you're relatively new to Internet Telephony and VoIP, then it may come as a bit of a surprise when CallerID for incoming calls shows the phone number for both the name and number of the incoming caller or when names that popped up using your plain old telephone's CallerID service no longer do. The problem is that most telephone providers only deliver a CallerID number when sending calls. Thus it is left to your service provider to look up the incoming number in a directory and supply the matching name. To put it another way, CallerID numbers are pushed to recipients, but CallerID names must be pulled from in-house databases. With Ma Bell and siblings, this was easy because they had all of the records. With some Internet Telephony Hosting Providers, it's a different story. The reverse is also problematic. Even though you may provide a CallerID name and number, most telephone companies throw the supplied CallerID name in the bit bucket and do their own lookups. So... if you're not in their directory, your number or nothing will be supplied instead of your actual name. Just another last vestige of monopoly preservation. With Asterisk, the simplest solution to this mess is to do your own lookups. And today we have an updated CallerID directory lookup service to assist: the CallerID Superfecta.

We have a second utility as well. Since moving to PBX in a Flash and Asterisk 1.4 and 1.6, we haven't provided a simple way to block or screen anonymous calls, i.e. those that show up in CallerID as either blank, anonymous, unknown, private, restricted, or toll free number. As they say, those are the usual suspects when it comes to weeding out unwanted callers. And today we'll provide several solutions from which you can choose. Our personal preference is to never answer these calls and route them straight to voicemail. You may be more curious than we are so we'll show you an option to screen calls using the Asterisk Parking Lot feature. Still others may hate these callers so much that you send them into IVR hell for hours at a time. And we've got some suggestions on that one, too.

Introducing CallerID Superfecta. We've dusted off an oldie but goodie today and reworked it a bit for newer versions of PHP. We also want to thank taiter and M Joyner for their whitepages.ca contribution to what used to be our CallerID Trifecta. The CallerID Superfecta now lets you choose up to four CallerID lookup sources for your incoming calls. The sources include the Google Phonebook, AnyWho, WhitePages, and our very own AsteriDex address book and robodialer. Complete installation instructions are available from our Best of Nerd Vittles site.

Installation and setup is a snap on all of the FreePBX-based aggregations including PBX in a Flash, Elastix, and trixbox. First, download and unzip the callerid.zip file into the root directory of the web server on your Asterisk system. Next, configure the sources you wish to use in callerid.php by setting the desired sources to 1 instead of 0 on the second page of the file. Then define the new CallerID Lookup Source in FreePBX. And finally, select the CallerID Superfecta as the lookup source for each of your Inbound Routes. The whole setup should take you less than two minutes. Now sit back and enjoy a much enhanced CallerID experience when incoming calls arrive on your Asterisk server.

Introducing the Creep Detector. Well, not so fast. The CallerID Superfecta doesn't get rid of the creeps that call wanting to sell you something or urging you to vote for your favorite Coroner. For that, you'll need a couple of tools. FreePBX includes an excellent web-based implementation of the Asterisk Blacklist. It allows you to specify the phone numbers of calls that should be blocked. You also can do this with a phone on your system by dialing *32 to blacklist the last caller or *30 to blacklist a specific phone number.

But, what about blacklisting all of those anonymous callers. Well, there's not an existing function in FreePBX to do it. Our preferred method goes like this. When an incoming call arrives, a message plays saying "Thanks for calling the Mundy's. Please hold a moment while I connect your call." During this message, a Stealth AutoAttendant will allow family members to press various buttons to be connected to various extensions. See the previous article for details. Once the IVR times out (in about 5 seconds), the call is passed to a Privacy Checker which screens the calls for creeps. If the call isn't identified as such, it is sent to a ring group. If a creep is detected, the system first plays a message that says: "Press 8 to be connected." If no key is pressed, we hang up. If 8 is pressed, the call goes to voicemail 704. If 4 is pressed, the call is passed to the ring group. This lets friends calling from phones with CallerID blocked still get through the maze.

So here's how to get it installed and working. Log into your server as root and add the following code snippet to the bottom of /etc/asterisk/extensions_custom.conf:

[custom-privacy-check]
exten => s,1,SetMusicOnHold(default)
exten => s,2,GotoIf($["${CALLERID(num)}" = ""]?s,16)
exten => s,3,GotoIf($["foo${CALLERID(num)}" = "foo"]?s,16)
exten => s,4,GotoIf($["${CALLERID(name):1:8}" = "nonymous"]?s,16)
exten => s,5,GotoIf($["${CALLERID(name):1:6}" = "nknown"]?s,16)
exten => s,6,GotoIf($["${CALLERID(num):1:6}" = "rivate"]?s,16)
exten => s,7,GotoIf($["${CALLERID(name):1:6}" = "rivate"]?s,16)
exten => s,8,GotoIf($["${CALLERID(num):1:9}" = "estricted"]?s,16)
exten => s,9,GotoIf($["${CALLERID(num):0:4}" = "PSTN"]?s,16)
exten => s,10,GotoIf($["${CALLERID(num):1:3}" = "oll"]?s,16)
exten => s,11,GotoIf($["${CALLERID(name):1:2}" = "--"]?s,16)
exten => s,12,GotoIf($["${CALLERID(name):0:1}" = ","]?s,16)
exten => s,13,GotoIf($["${CALLERID(name):1:3}" = "oll"]?s,16)
exten => s,14,GotoIf($["${CALLERID(num):0:3}" = "000"]?s,16)
exten => s,15,Dial(Local/777@from-internal)
exten => s,16,Playback(custom/nv-press8)
exten => s,17,Set(TIMEOUT(digit)=10)
exten => s,18,WaitExten(10)
exten => s,19,Hangup
exten => 4,1,Dial(Local/777@from-internal)
exten => 4,2,Hangup
exten => 8,1,VoiceMail(704@default)
exten => 8,2,Hangup

You'll need to make a couple of changes in the code above before using it. In lines s,14 and 4,1, modify extension 777 to reflect an extension or ring group on your phone system that you want to call after incoming calls are screened for creeps. In line 8,1, modify 704 to reflect a voicemail box that is active on your system and that should be used for recording messages from unwanted callers.

The next step is to add the "Press 8 to be connected" message to your system. While still logged in as root, issue the following commands:

cd /var/lib/asterisk/sounds/custom
wget http://bestof.nerdvittles.com/applications/callerid/nv-press8.wav
chown asterisk:asterisk nv-press8.wav
chmod +x nv-press8.wav


Now we need to configure your FreePBX setup to use the code above. The easiest way is to modify your Stealth AutoAttendant IVR and simply change the timeout destination (t) to a Custom App: custom-privacy-check,s,1. Save your changes and reload your dialplan, and you're all set.

Some additional ideas have also been floated on the PBX in a Flash Forum for handling anonymous callers. If you'd prefer to park these calls and announce them, see this thread. And here's an embellished version that gives you options to accept the call, send it to voicemail, or banish the caller. Enjoy!

---

Vitelity Special: Time Is Running Out. Remember, you only have one more week until July 15 to get your half-price DIDs and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the PBX in a Flash project gets a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up before the end of June, and you can purchase a Tier A DID with unlimited incoming calls for just $3.79 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S.

The one thing that Mark Spencer and I totally agree upon is that neither one of us shouts "Roll Tide" very often. So... my apologies to all of my Auburn friends for the headline. I didn't really mean it! We'll have more to say about Digium's Pit Bull approach to software development in coming weeks. But today we're celebrating the arrival of summer and Joe Roper's safe passage through the Straits of Gibraltar. We'll leave Asterisk politics for a gloomy, rainy day. From the Never to Be Outdone Department: If you need your weekly fix of The Worst of Asterisk, then here's a link to the FreePBX news item describing Fonality's latest shenanigans. You might want to start with the initial revelations and then hold your nose while reading the follow-up responses. Pretty sad stuff for an open source project or any other business for that matter. The Silver Lining: Could there be a better mousetrap looming on the horizon? Stay tuned!

Introducing xTide for Asterisk. Whether you're a beach bum or would just love to get a current surf report from the nearest telephone, then today's your lucky day! With xTide for Asterisk, you get the latest text-to-speech (TTS) tide predictions as well as solar and lunar forecasts for any port of call in the United States. To use the application, just dial T-I-D-E from any phone connected to your PBX in a Flash system and Allison or Egor will tell you everything you ever wanted to know about sunrise, sunset, moonrise, moonset, and today's and tomorrow's high and low tide predictions for almost any oceanfront location. And we'll also show you how to connect the system up to a phone number of your choice so that tide reports are as close as your cellphone. Of course, retrieving reports with Flite and Egor is free. Cepstral with Allison will set you back a meager $25. If you want a quick demo, dial 425-906-5918. Thanks, IPKall.

Overview. The xTide for Asterisk application is a little different than some of our other text-to-speech applications which rely upon servers hosted by other folks. With xTide for Asterisk, we're actually going to install the full xTide app (about 40MB of code) on your PBX in a Flash server. As configured, it's a very well-behaved application that uses virtually no system resources except when you retrieve a report. And, remember, xTide should not be used for navigation or anything else that really matters! It's a statistical application, not a weather expert nor a rocket scientist. But special thanks anyway to David Flater for his continuing development of xTide! With xTide for Asterisk, you can generate text-to-speech reports over the phone, and you also can access David's complete xTide application by pointing your web browser to the dedicated xTide web server to retrieve anything you ever wanted to know about solar, lunar, and tide forecasts for almost anywhere in the United States. There's even an x-Windows interface if you've installed it on your system. The complete list of locations supported by xTide is available here or you can retrieve the list from your own xTide server: http://serverIPaddress:88/index.html. And, yes, for our foreign friends, there's a worldwide version of the harmonics files which is available for non-commercial use only. You'll need to manually install it if you want to use it. Source RPMs also are available if you want them.

Prerequisites. We've tested this installation with PBX in a Flash 1.1 and 1.2 so it's a one-minute drill if you've already got one of those systems in place. You can download the current version of PBX in a Flash here. And complete documentation is available here. With other Asterisk-based aggregations, there may be some additional wrinkles, but we'll leave those for the pioneers to iron out. For PBX in a Flash users, you simply download our script, mark it as executable, and run it. A minute later, you're ready to begin your surfing career.

Setup Tips. Once you complete the installation, you'll have three separate xTide applications available. xtide is actually the x-Windows interface which we're not going to be using because of the x-Windows overhead on your server. tide is the command-line equivalent which we'll be using to create TTS reports for Asterisk. You also can use it interactively, and we'll show you how in a minute. There's also a a web server dedicated to serving up all sorts of xTide reports which can be retrieved with your favorite browser. While it is activated after the install completes, it won't automatically restart when you reboot your server unless you set it up to do so. We did this to minimize the load on your system if you only plan to retrieve reports using the TTS telephony interface. The only other setup step involves choosing your default location for the reports, and there are pages of locations from which to choose. We'll walk you through the location setup process momentarily, but first, let's install the application.

Installation on PBX in a Flash Systems. We've tested this installation on dedicated PBX in a Flash 1.1 and 1.2 servers. It does not run on hosted servers without satisfying additional dependencies which we'll leave to the hosted system providers to work out. To install, log into your server as root and issue the following commands:

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

If you plan to use the application with the Cepstral TTS engine, choose option 2 when prompted. Note: We recommend you first install Cepstral before proceeding. If you're not using Cepstral, choose option 1 for the default Flite TTS engine. But keep in mind that Flite does not work with Asterisk 1.6-beta while Cepstral does... if you follow our tutorial.

Choosing a Default Location. Tides, sunrise, and sunset obviously differ depending upon your coastal location. As installed, you'll get the information for Pawleys Island, South Carolina which is the site of our webcam. If you'd like to try out our system, just dial our free IPKall number for the Pawleys Island xTide report: 425-906-5918. To use another location on your own system, you first need to decipher whether it's in the list of supported locations. Once you have your location, test it using the following command: tide -l "sitename". For example, tide -l "pawleys" returns the tide information in the default setup. Once you get a report for the desired location, edit /etc/asterisk/xtide.conf and replace "pawleys" with the SITE location you've chosen. Then add a descriptive SITENAME. Do NOT add any blank lines to the config file, or nothing will work. Save your changes and run a simple test by dialing T-I-D-E from a telephone connected to your server.

Using the Web Interface. To use the web interface on your private network, go to the following link using the actual IP address of your Asterisk system: http://serverIPaddress:88/. Complete documentation and a FAQ are available. To access the web interface from the Internet, you'll need to redirect TCP port 88 to the private IP address of your Asterisk server. When you reboot your server, the dedicated web server for xTide will not start automatically. We set it up this way for those that don't want the overhead of a third web server. Remember Apache and WebMin already are running. If you don't plan to use the xTide web interface, then you're all set. If you want to permanently enable xttpd, then log into your server as root and issue the following command:

chkconfig --levels 2345 xttpd on


Creating an External Phone Number for xTide. In a previous column, we walked you through setting up a SIP proxy with PBX in a Flash. To expose xTide for Asterisk to the phones of the outside world, the cheapest approach is a free phone number from IPKall. Before obtaining one, you'll need a dyndns.org fully-qualified domain name that points to the public IP address of your server. Then set up your SIP proxy for xtide and create an Inbound Route in FreePBX for xtide that routes incoming calls to custom-xtide,s,1. Now insert the following code at the bottom of /etc/asterisk/extensions_custom.conf and reload your dialplan. Here's the code:

[custom-xtide]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,Dial(local/8433@from-internal)
exten => s,4,Hangup


Now sign up for your phone number at IPKall and use a valid email address. Otherwise, you'll never receive your phone number! Your IPKall entries should look something like this:

Account Type: SIP
SIP Phone Number: xtide
SIP Proxy: yourdomain.dyndns.org
Email Address: yourname@gmail.com
Password: yourpassword
Seconds to Ring: 120


Programmer's Corner. Except for a few lines of dialplan code, xTide for Asterisk makes exclusive use of bash commands to manipulate the output of tide and turn it into speech. The results are stored in /tmp so Linux will handle file cleanup automatically. The application that does the heavy lifting is /var/lib/asterisk/agi-bin/xtide. You'll also find xtide.flite and xtide.cepstral in the same directory. Should you wish to change text-to-speech engines, just copy the appropriate file over to xtide. For the whiz kids, we're always open to suggestions on how to improve the code which makes extensive use of eval, sed, and cut to get the proper results from the xtide output. No need to elaborate on our crappy bash programming skills. We already know that!

One piece of code we've included to reduce the overhead on Cepstral usage is a test for the timestamp of /tmp/xtide.txt. If it matches today's date, then we simply skip processing of a new xtide request and play the previously generated xtide.wav file. This might matter if you decide to change TTS engines in the middle of the day. To force a regeneration of the xtide.wav file, just rm -f /tmp/xtide.* after making a change in your xTide TTS engine. One final tip... if you generate your own TTS files using tide, don't use our xtide.* and other temporary file names (which start with the name of your chosen port city) while logged in as root, or this application will be unable to delete or overwrite them. HINT: su asterisk before playing. Enjoy!

Other TTS Applications for Asterisk. xTide for Asterisk is the latest in our growing collection of text-to-speech applications for Asterisk:

• AsteriDex RoboDialer
• Asterisk Weather Station by Airport Code
• Asterisk Weather Station by Zip Code
• Worldwide Weather Forecasts
• MailCall for Asterisk
• NewsClips for Asterisk
• TeleYapper
• Telephone Reminders for Asterisk
• CallWho for AsteriDex (beta)

To download the latest versions, visit our Best of Nerd Vittles site.

---

Vitelity Special: Time Is Running Out. Remember, you only have two more weeks until July 15 to get your half-price DIDs and 60 free minutes from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the PBX in a Flash project gets a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up before the end of June, and you can purchase a Tier A DID with unlimited incoming calls for just $3.79 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S.

We began our 15-minute adventure last week with a turnkey install of Asterisk onto a $199 Everex gPC2 using a fully-customized version of PBX in a Flash. If you haven't yet read the first article, start there. Today we want to cover what components are included and walk you through using most of them. When we're finished, you'll have a good idea why PBX in a Flash is not only different but also a quantum leap forward in the turnkey IP telephony marketplace. We'll also cover adding RAID 1 redundant drive support to your new server for about $40.

Putting Your Backup System Into Operation. Hopefully, you heeded our recommendation and purchased a $20 4GB USB flash drive to store backups of your new PBX in a Flash system. Cheap insurance! Now let's put it into production. In the /root folder of your new system, you'll find a PDF with complete documentation for the new Mondo Rescue backup system. If you flip to Appendix A, it will walk you through formatting your new flash drive for use with the backup software. Basically, you're going to delete the existing partitions on the drive, repartition it as a FAT32 partition (don't use the existing partition even if it says FAT32!!), and then reformat the drive. It takes under a minute to do it all. Once you're finished, let's initiate a backup just to be sure everything is working. Log into your server as root and type /etc/cron.weekly/disk-backup.cron. When the command prompt returns in about 30 minutes, type /root/usbcheck.sh to get a listing of the files on your USB flash drive. Now you can sit back and relax knowing that every Sunday night a new full system backup will be loaded onto your flash drive. Should something go horribly wrong with your main drive down the road, it's a simple matter to burn CDs of the ISO backup and reload everything, the same process you used to build your new system in the first place. Remember, we provided you a Mondo Rescue backup to build your system from ours so you know it works. For us at least, having automatic backups of your data is a critical component in any computer system, particularly your entire telephone system. While Asterisk aggregations are a dime a dozen these days, no one else has implemented any backup solution except PBX in a Flash.

Text-to-Speech on Steroids. The next thing you need to do is install Cepstral with Allison on your system. This gives something close to perfect text-to-speech capability for your entire phone system for under $25. And, yes, you can try it out first without spending a dime. Log into your server as root and type install-cepstral. Accept the defaults except create the missing directory when prompted. You're done. That was hard wasn't it. We'll test it out in a few minutes.

PiaF Software Update Service. At least for now, the PBX in a Flash Software Update Service continues to be a free option on all PBX in a Flash systems so by all means use it to keep your system current, bug-free, and secure. Log into your server as root and type update-scripts. Once the new scripts are loaded onto your system, type update-fixes. Yes, you can build an Asterisk system from many other ISO distributions. But you won't find another one that can keep your system current and secure without starting all over with a new ISO install. And when you want the latest and greatest version of Asterisk without missing a beat, that's easy, too. Just type update-source and have a cup of coffee while your system is upgraded. And don't forget to run update-fixes one more time to clean up any mess created by the upgrade.

Help at Your Fingertips. And, what if you forget all of these commands down the road and you're too lazy to pull out the documentation? Not to worry! Log into your server as root and type help-pbx.

What's Next? Now that you have a stable, secure, and up-to-date server, let's have some fun. We've loaded and preconfigured most of the Nerd Vittles applications in this build so all you have to do is learn the numbers to dial to use most of the applications. Here's a quick thumbnail sketch for each of the applications:

• AsteriDex RoboDialer and Telephone Directory
  This app gets you a phonebook, a web-based dialer using a browser or your cellphone, and a CallerID lookup source when used in conjunction with Ultimate CNAM. To add and update entries or lookup numbers, point your web browser to the IP address of your server: http://ipaddress/asteridex4/. For cell phone access, point the web browser on your cellphone to the public IP address or fully-qualified domain name of your server: http://publicIPaddress/cellphone/. Click on the link above for complete documentation and security suggestions.
• Telephone Reminders 4.0 with Support for Recurring Reminders and Web-based TTS Reminder Messages
  This app lets you schedule reminders for future events by telephone (dial 1-2-3) or with a web browser (http://ipaddress/reminders/). When the appointed date and time arrives, Asterisk swings into action and places a call to the number you designate to deliver a customized reminder message. Recurring reminders (daily, weekday, weekly, monthly, and annual) also are supported. And the text-to-speech web interface lets you schedule and deliver reminders using either Flite or Cepstral-generated messages with any web browser. For more info, click on the link above.
• NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS) - Dial 5-1-1
• Weather Reports by Airport Code (TTS) - Dial 6-1-1
• Weather Reports by ZIP Code (TTS) - Dial Z-I-P
• Worldwide Weather Forecasts (TTS) - Dial 6-1-2
• MailCall for Asterisk: Get Your Email By Telephone (TTS)
  This app reads your emails to you over the telephone. Some setup is required to plug in information about your email account. Once configured, dial 5-5-5 to retrieve your messages. Click on the link above for setup instructions.
• TeleYapper 4.0 Message Broadcasting System - Dial M-S-G (licensed for non-commercial use only!)
• CallWho for TTS Retrieval and Dialing of Entries in the AsteriDex Database (TTS)
  After entering contacts in AsteriDex, run http://serverIPaddress/asteridex4/dialcode.php to populate the dialcodes. Then dial 4-1-2 and enter the first three letters of anyone in your AsteriDex database to place a call.
• TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones - See setup instructions in last week's article

Of course, there are literally hundreds of things you can do with your PBX in addition to running the Nerd Vittles applications. Here's a short list of some of our favorites with some tips to get you started. The best source of information for more detail is our original article on PBX in a Flash 1.2 and the PBX in a Flash Forum.

• Stealth AutoAttendant with Welcome and Application IVRs
  Whenever an incoming call comes into your PBX, a generic greeting will play. If no button is pressed on the caller's phone, the call then will be routed to a ring group (700) for all of the extensions set up in that ring group. If no one answers, the call will be sent to the voicemail box for extension 701. While the greeting message is playing, the caller can press a digit on their phone to activate a hidden option in the Main IVR. As delivered, the only one that works is 0. This presents the caller with a list of Nerd Vittles apps from which to choose. You can add other options by modifying the Main IVR settings in FreePBX. To try out the Main IVR from any extension on your system, dial 7-7-7.
• Key Telephone Support Using Park and Parking Lot
  Most PBXs do not support shared line appearances like the old key telephones from Ma Bell. With these phones you could answer a call, place it on hold, and then someone else could pick up the call by pressing the blinking light on their phone. Our Aastra phone setup does much the same thing except, instead of placing a call on hold, you press the Park button. The parked extension number then will be read to you by Allison (starting with 71). Anyone else on your system can retrieve the parked call by pressing the ParkLot button on their Aastra phone and selecting the call to be retrieved by CallerID. Or, if the recipient knows the parking lot extension (e.g. 71), the recipient can pick up any phone and dial that extension number to retrieve the call.
• Intercom/Paging Support
  The Aastra phone setup for PBX in a Flash fully supports intercom calls and paging by pressing the ICom button on the phone. For more information, click Setup, Paging and Intercom from within the FreePBX web interface.
• Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
  Your system is preconfigured to support a USB Bluetooth dongle. No additional software installation is required. When properly configured, this lets you automatically forward your calls to your cellphone just by leaving your home or office with your Bluetooth-enabled cellphone. When you return, your calls will magically begin ringing on your local extension again. Click the link above for setup instructions.
• DISA
  Direct Inward System Access lets you call into your PBX and get dialtone to make an outbound call. To use it, you typically would add it as a hidden option on your IVR with a very secure password. We have preconfigured DISA support on your server. Just be sure you change the password to something very secure before activating it. To change the password, click Setup, DISA, DISAmain in FreePBX. Then save your changes and reload the Asterisk dialplan.
• Blacklisting with Web and Telephony Interfaces
  To block future calls from the last person who called you, dial *32. To block calls from a specific phone number, dial *30. To remove a number from the blacklist, dial *31. You also can use FreePBX to blacklist certain numbers. Just click Setup, Blacklist to access the web interface.
• CallerID Name Lookups from 8 Providers
  Most telephony providers reliably pass CallerID numbers but discard CallerID name info. With Ultimate CNAM which is preinstalled on your system, you can look up CallerID names from up to 8 different directory providers. To activate it, use FreePBX and click Setup, Inbound Routes, DefaultIncoming. Scroll down to CID Lookup Source and choose Ultimate CNAM from the dropdown box. Save your changes and reload the dialplan. For complete documentation, consult cnam_user_guide.pdf in the /root folder on your server. To choose the providers to use for the lookups, log into your server as root and type: cnam-config.pl
• Weekly Automated System Backups to a Flash Drive
  See the first section of today's article for the one-minute setup instructions.
• One Touch Day/Night Service
  With our Aastra phone setup, there is a DayNite button that toggles your system between Day and Night operation. As configured, the Night option transfers all calls to voicemail for extension 701. The Day option routes all incoming calls through the Main IVR which routes calls to the 700 Ring Group on timeout. To activate Night service from an Aastra phone, just press the DayNite button. To deactivate Night service, just press the button again. You also can dial *28 from any phone on your system to toggle Day/Night mode.
• Music on Hold
  Royalty-free music on hold is provided as part of the basic Asterisk install. Additional music can be added through the Music on Hold option in FreePBX. WAV files must be PCM Encoded, 16 Bits, at 8000Hz. See this thread for assistance. For other royalty-free and free music on hold, start here, choose Creative Commons for the License Type, and then click Go.
• Voicemail with Email Delivery of Messages and Pager Notification
  All of these settings are performed within FreePBX for each extension. Choose Setup, Extensions, and pick one of the extensions you already have created. Make certain that Voicemail Status is enabled. Then enter a valid email address and pager address. To include the voicemail as an attachment in the delivered email message, set Email Attachment to Yes. To include the CallerID in the voicemail message, set Play CID to Yes. To include the date and time of the call, set Play Envelope to Yes. To delete the voicemail message from the system after emailing it, set Delete Vmail to Yes. Don't ever do this until you're sure it's working reliably! If you want the option of calling back the caller when you retrieve your voicemail message by phone, set VMoptions to callback=from-internal. Submit your changes and reload the Asterisk dialplan to put the modifications into effect. If the emails are not delivered, then it may be because your ISP is blocking downstream SMTP traffic. To reconfigure your server to use gMail or Comcast as your SMTP host, click on one of the links.
• Voicemail Blasting
  This feature allows you to record a message and distribute it via voicemail to one or more extensions without actually calling the users. We've already configured extension 500 to send voicemail blasts to extensions 701 and 702. You can adjust the destinations in FreePBX by choosing Setup, Voicemail Blasts, Vmail (500). You also can add additional extensions to handle voicemail blasts to a different group of phones.
• Visual Voicemail for Asterisk with Aastra phones and the Apple iPhone
  The Aastra phone setup provided in this build includes a button for Visual Voicemail. Each call is displayed with its CallerID and the date and time of the call. Scroll down the list and select the voicemail message you wish to play. You will be prompted for your voicemail account password and then the message will play. You also can delete messages using this interface. There's also a Web 2.0 for iPhones that provides Visual Voicemail for any Asterisk voicemail box. Using the external IP address or fully-qualified domain name of your server, point Safari on your iPhone to http://externalIPaddress/iphone/ and enter your extension number and voicemail password.
• Cell Phone Direct Dial
  There are two ways to make a cellphone an integral part of your PBX. The first involves setting up a specific extension for each cellphone and forwarding incoming calls to that extension to your cellphone. We've created extension 501 to show you how it's done. Once the extension is created, simply log into your server as root and issue a command like this where 6781234567 is your actual cellphone number:
  
asterisk -rx "database put CF 501 6781234567"

  When callers dial 501 on your system, your cellphone will automatically ring. Another option is to use FreePBX's Follow Me function under Setup. With this option, you can specify multiple destinations for incoming calls to a specific extension. Point to the Ring Strategy option and review the available choices. Choose the one that best meets your needs. Then enter the numbers to be called. Numbers outside your PBX should be in the format 6781234567# and must match your outbound dialing rules. You also can choose the time to attempt the call and what to do if no one answers. Very slick!
  
• Call Forward: All, Busy, No Answer
  While you can certainly use FreePBX's Follow Me functionality to accomplish any flavor of call forwarding, you also can dial codes from any extensions to activate call forwarding. To activate Call Forwarding All, dial *72; for Call Forwarding Busy, dial *90; for Call Forwarding No Answer, dial *52. To deactivate Call Forwarding All, dial *73; for Call Forwarding Busy, dial *91; for Call Forwarding No Answer, dial *53. To deactivate Call Forwarding All from a different extension, dial *74; for Call Forwarding Busy, dial *92. You also can activate and deactivate Call Forwarding from any Aastra phone using our default setup.
• Call Waiting
  To activate Call Waiting from any extension (which is the default), dial *70. To deactivate Call Waiting, dial *71.
• Call Pickup
  To pickup a call ringing on another extension, dial **.
• Zap Barge
  To barge into an existing call, dial 888.
• Call Transfer: Attended and Blind
  For attended call transfers where you can remain on the line until the other party answers, dial *2. For unattended call transfers, dial ## and then the number to which the call should be transferred.
• Dictation Service with Email Delivery
  Before using FreePBX's dictation service, you must activate Dictation Services for the specific extension to be used. Using FreePBX, go to Setup, Extensions, and click on the desired extension. Scroll down to the Dictation Services section of the form and enter your email address, the format of the sound files to be used, and change Dictation Service to Enabled. Save your settings and reload the dialplan. Then you can dictate your message by dialing *34. Once you finish your dictation, you can email it to your email address for this extension by dialing *35.
• Do Not Disturb
  To activate Do Not Disturb on any extension, dial *78. To deactivate Do Not Disturb, dial *79. There's also a button to accomplish the same thing with our Aastra phone setup.
• Phonebook Dial by Name - Dial 4-1-1
• VoiceMail Options
  To retrieve your voicemail from any phone, dial *97. To retrieve voicemail for a different extension, dial *98 or *98701 where 701 is the extension desired. To leave a voicemail message for any extension with voicemail enabled, dial *701 where 701 is the extension desired.
• Speed Dial
  To set up a user speed dial entry, dial *75. To call any previously established speed dial entry, dial *0 plus the speed dial number. To create or modify speed dial entries in FreePBX, click Tools, Asterisk Phonebook. You also can import entries from a CSV-formatted file.
• Flite and Cepstral Text to Speech (TTS)
  Flite TTS is installed by default with all PBX in a Flash systems using Asterisk 1.4. The Asterisk developers have broken Flite in the Asterisk 1.6-beta and refuse to fix it. Cepstral can be installed in both Asterisk 1.4 and 1.6-beta by logging in as root and typing install-cepstral. To use Flite with Egor in your dialplan, here's the syntax:
  
exten => 444,5,Flite("Hello World.")

  To use Cepstral with Allison in your dialplan, use this syntax:
  
exten => 444,5,Swift("Hello World.")

• One-Click Cepstral TTS Install with Allison
  Just Type install-cepstral to install Cepstral. For detailed instructions on reconfiguring Nerd Vittles apps to use Cepstral instead of Flite, see this article. No software needs to be reinstalled. Simply change the dialplan and PHP app settings to use Cepstral as explained in the article. For more background on Cepstral, read this article. To register your newly installed Allison voice, go to this link. Be sure you select U.S. English language, Allison-8kHz voice, and Linux platform before you check out, or it's money down the drain. For 20% off your Cepstral license registration, use promo code “REALLUSIONTTS” when you check out. Write down the name, company (optional), and key that is issued once you fill in the blanks. Then log into your server as root, and type swift --reg-voice. Fill in the blanks with the information you wrote down above, and you're all set.
• Windows Networking with SAMBA
  Windows Networking with SAMBA is enabled by default in this special build. The default workgroup is "workgroup." To change the workgroup, log into your server as root and edit /etc/samba/smb.conf. Then restart SAMBA: service smb restart. You then can connect to your server from any computer that supports Windows networking using root as your username and whatever root password you created.
• Linux Firewall
  The IPtables firewall is enabled by default in all PBX in a Flash systems. For this build, we have enabled SAMBA access to your server. To disable it, log into your server as root, and issue the following commands:
  
cp /etc/sysconfig/iptables.nosamba /etc/sysconfig/iptables
service iptables restart

• WebMin
  WebMin is often described as the Swiss Army Knife of Linux. It provides a terrific web interface to Linux.everything. It is enabled by default in this install. To access it using a web browser, go to http://serverIPaddress:9001/ and login as root with the password you set up above for WebMin access. For complete documentation, go here.
• PBX in a Flash Software Update Service To Keep Your System Current
  To load current fixes for this build of PBX in a Flash, log into your server as root and type the following commands:
  
update-scripts
update-fixes

  To upgrade your system to the latest version of Asterisk 1.4, log into your server as root and type the following commands:
  
update-scripts
update-source
update-fixes


---

More Good News with the Everex gPC2. From the "Learn Something New Every Day Department," this news just in. The Everex gPC2 has built in hardware SATA RAID 1 support that actually works. What you'll need to get this going is a second 80GB hard disk to match the one delivered in your original box. Total cost: about $40. If one disk fails, the other kicks in automatically. Here's a link to purchase your drive. And here's the link that'll tell you how to get everything set up. Before you begin, make certain that you have a current ISO backup on your flash drive so that you can restore your system once the RAID setup is up and running. See the top of this article for the backup and testing procedure.

---

Vitelity: The Best Provider and Pricing on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the PBX in a Flash project gets a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up before the end of June, and you can purchase a Tier A DID with unlimited incoming calls for just $3.79 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price OR the call quality! Trust us. We've tried just about everybody. Update: This offer has been extended until July 15.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good Vitelity can be, pick up a phone on your system and dial 678-444-2445. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications using Vitelity as the inbound DID provider. Now hang up and place another call to the same server by dialing D-E-M-O which makes a free SIP-to-SIP connection between your new server and our server at Aretta Communications. Enjoy!

Well, okay. We confess that today's creation doesn't quite measure up to the legendary Orgasmatron... but, look out Woody Allen, we're close. It's been a couple of years since we released our first preconfigured, turnkey Asterisk install. Much has changed both in Asterisk and in the hardware and software environment since 2006. So today, to celebrate the six month anniversary of PBX in a Flash and the brand new PBX in a Flash 1.2 release, we're taking another stab at it. From the time you insert the CD 'til you have a functioning Asterisk PBX with all the bells and whistles imaginable... 15 minutes!

Our approach today is a little different than the last time around. The processing overhead of CentOS 5.1 has made VMware problematic. Luckily, the price of hardware has dropped like a rock. So today we're comfortable recommending the best phone, the best PC, and the best provider on the planet. And you'll still have your arms and legs intact after you pay the piper. If you've been following along with our articles, you already know that we've identified what we believe to be the perfect Asterisk SIP phone, the Aastra 57i, and we've also identified a perfect small business/home computer on which to run a production Asterisk server for about 50 employees, the Everex gPC (aka "The WalMart Special"). Now that the second generation Everex gPC2 is readily available, we decided to preconfigure one of these systems from the ground up and then make a 2-disk ISO image backup of the whole system using Mondo. So, once you download the ISO images and burn your CDs, it's a 15-minute No-Brainer to install the entire image onto your own Everex gPC2. But you must have a gPC2 for this to work so accept no substitutes, and don't try this with any other hardware or you'll end up with an Electronic Brick instead of an Orgasmatron. The $199 gPC2 systems are available from WalMart and NewEgg among others.

We've preconfigured outbound and incoming trunks from some terrific providers as well as some extensions on your new system. So you literally can sign up for service with these providers, plug in your phones, and you can be in full operation in under an hour. Our only word of caution is not to use these ISO images on another type of computer. Everything has been specifically tailored to the stock gPC2 and chances are very good that the install would not proceed much past erasing and reformatting your hard disk on a different flavor machine. We also recommend that, if you want to add our recommended $25 extra gig of RAM to the gPC2, hold off on installing it until after you have loaded the new ISO image. So... what do you get with this preconfigured build?

In addition to all of the goodness of a stock PBX in a Flash 1.2 build including Asterisk 1.4 running under CentOS 5.1 with all the latest and greatest versions of FreePBX, Apache, MySQL, and PHP, you also get 10 preconfigured Nerd Vittles applications for openers:

• AsteriDex RoboDialer and Telephone Directory
• Telephone Reminders with Support for Recurring Reminders and Web-based TTS Reminder Messages
• NewsClips for Asterisk featuring Dozens of Yahoo News Feeds (TTS)
• Weather Reports by Airport Code (TTS)
• Weather Reports by ZIP Code (TTS)
• Worldwide Weather Forecasts (TTS)
• MailCall for Asterisk: Get Your Email By Telephone (TTS)
• TeleYapper 4.0 Message Broadcasting System
• CallWho for TTS Retrieval and Dialing of Entries in the AsteriDex Database (TTS)
• TFTP Server with preconfigured setups for 15 Aastra 57i SIP telephones

In addition, you get dozens of preconfigured telephony applications and functions that would take even an expert the better part of a year or two to build independently. And, unlike all of the other distributions, we build Asterisk from source so it's simple to modify and upgrade whenever you feel the need. Here's a short list of what you have to look forward to:

• Stealth AutoAttendant with Welcome and Application IVRs
• Key Telephone Support Using Park and Parking Lot
• Intercom/Paging Support
• Bluetooth Proximity Detection with Automatic Call Forwarding to Cell Phone
• DISA
• Blacklisting with Web and Telephony Interfaces
• CallerID Name Lookups from 8 Providers
• Weekly Automated System Backups to a Flash Drive
• One Touch Day/Night Service
• Music on Hold
• Voicemail with Email Delivery of Messages and Pager Notification
• Voicemail Blasting
• Visual Voicemail for Asterisk with Aastra phones and the Apple iPhone
• Cell Phone Direct Dial
• Call Forward: All, Busy, No Answer
• Call Waiting
• Call Pickup
• Zap Barge
• Call Transfer: Attended and Blind
• Dictation Service with Email Delivery
• Do Not Disturb
• Gabcast
• Phonebook Dial by Name
• Speed Dial
• Flite Text to Speech (TTS)
• Windows Networking with SAMBA
• Linux Firewall
• PBX in a Flash Software Update Service To Keep Your System Current
• One-Click Cepstral TTS Install with Allison... Just Type install-cepstral

Prerequisites. As mentioned, you'll need a $199 Everex gPC2 (WalMart or NewEgg ) to use this build. We also recommend an additional $25 gig of RAM for anything other than home use. We also recommend a 4GB USB flash drive on which to store automatic weekly backups of your new system. Finally, you'll need to cough up a whopping $5 to download the two-disk ISO image for this build. And, yes, we eat our own dog food. The ISO images you'll be downloading were captured as a backup on the flash drive of our gPC2 lab machine. If you use this special build, it seemed only fair that you cover the cost of the bandwidth to download it. As most of you know, we don't have the luxury of freeloading off SourceForge for our downloads. And we didn't want to impose upon our existing bandwidth providers to bring you this custom image. The good news is that, once you download the image from DreamHost, you are more than welcome to pass it along to one or more of your friends or business acquaintances at no charge. You can even do it electronically through the DreamHost Files Forever program. And, if you'd like to host this image for your fellow man at no cost, be our guest... and thank you! Bottom line: For about $250, you'll have the slickest, most reliable PBX on the planet with rock-solid weekly backups and, of course, the one-of-a-kind PBX in a Flash Software Update Service!

Getting Started. Once you have purchased your Everex gPC2, take it out of the box, plug it into your LAN with DHCP and DNS support and Internet connectivity. Having said that, we strongly recommend that you always keep your system running behind a NAT-based firewall/router. Almost any home router will do. Don't redirect any ports to the machine and don't turn the PC on just yet.

Download the two ISO images for the gPC2 from here. If you don't know how to create a CD from an ISO image, read that section from our article last week. In fact, read the whole article. It'll help you immensely down the road. Once you have the two CDs in hand, turn on the gPC2 and quickly insert Disk 1 into the CD/DVD drive and close the drive. If you don't see a Mondo Rescue screen within a minute or less, turn the machine off and then back on again. At the Mondo Rescue main screen, type nuke and press the Enter key. This will erase, repartition, and reformat your hard disk in case you didn't know. This is normal. If you get any kind of errors about incorrect drive or partition names, halt the install by pressing CTL-ALT-DEL and remove the CD. You'll need to install PBX in a Flash using our standard ISO which is available here. Otherwise, go have a cup of coffee and come back in about 12 minutes. When prompted, insert Disk 2 and press the Enter key to finish the install. When the CD ejects, remove it and your gPC2 will reboot after you type exit.

After the reboot finishes, type root at the login prompt for your username and password for your password. The IP address assigned by your DHCP server should appear near the top of the screen. Write it down. If there is no IP address, your machine does not have network connectivity or access to a DHCP server with an available IP address. Correct the problem and reboot.

Securing Passwords. We're going to change five passwords now. For the time being (until you've done some reading), think up one really difficult password (that you won't forget) and use it for all five passwords. At the root@pbx:~ $ command prompt, type the following commands and type in your new password when prompted. Don't forget your password or you'll get to put in your two CDs and start over.

passwd
passwd-maint
passwd-wwwadmin
passwd-meetme
/usr/libexec/webmin/changepass.pl /etc/webmin root yournewpasswordhere


Now, using a web browser, go to the IP address of your new PBX in a Flash server. Click Administration. Log in as admin:password. Then click Menu Config. Change Admin Pwd to a new password that you're NOT using elsewhere. Now click Update and then Done. Click Administration again and then Asterisk Mgmt (FreePBX). If you're prompted for username and password, use admin:password for now. After FreePBX loads, click Setup and then Administrators. In the far right column, click admin, fill in your new password, and click Submit Changes. Then do the same thing for maint. Finally, click on the orange Apply Configuration Changes button and then Continue with Reload. Whew!

Don't change any other passwords without first contacting us. Regardless of what you may read elsewhere, PBX in a Flash is now secure. If you want more details, read this article and this thread.

Permanently Setting the IP Address. There are different schools of thought on whether to use a fixed or dynamic IP address. Most hardware-based routers support DHCP IP address reservations. The simplest way to permanently secure the existing IP address for your server is to reserve it on your router. If you'd prefer to assign your own IP address, we have included the deprecated netconfig utility which can be run after logging into your server as root. Sometimes you will need to run it once, enter your settings, reboot, and then repeat the drill. Then you should be all set.

Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone (highly recommended). Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device (the size of a pack if cigs) known as a Sipura SPA-1001. It's under $60. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. Once you get it, plug the SPA-1001 into your LAN, and then plug your phone instrument into the SPA-1001. Your router will hand out a private IP address for the SPA-1001 to talk on your network. You'll need the IP address of the SPA-1001 in order to configure it to work with Asterisk. After you connect the device to your network and a phone to the device, pick up the phone and dial ****. At the voice prompt, dial 110#. The Sipura will tell you its DHCP-assigned IP address. Write it down and then access the configuration utility by pointing your web browser to that IP address.

Once the configuration utility displays in your web browser, click Admin Login and then Advanced in the upper right corner of the web page. When the page reloads, click the Line1 tab. Scroll down the screen to the Proxy field in the Proxy and Registration section of the form. Type in the private IP address of your Asterisk system which you wrote down previously. Be sure the Register field is set to Yes and then move to the Subscriber Information section of the form. Assuming you're using the preconfigured extensions starting with 701, do the following. Enter House Phone as the Display Name. Enter 701 as the User ID. Enter 1234 as the Password, and set Use Auth ID to No. Click the Submit All Changes button and wait for your Sipura to reset. In the Line 1 Status section of the Info tab, your device should show that it's Registered. You're done. Pick up the phone and dial 1234# to test it out.

Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with freePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best phone out there is the Aastra 57i for under $200. Another $100 buys you the Aastra 57i CT with a cordless DECT phone.

Configuring Aastra 57i SIP Phones. Your new system comes preconfigured to automatically configure up to 15 Aastra 57i phones. Plug each phone into your network and wait for it to boot. Once it boots, press the Option button, then Phone Status (3), then IP & MAC Address (1). Write down each phone's IP address and MAC address. Then press Done to exit from the menus.

Next, we need to tell your phone to use your new server as the TFTP server to obtain its setup. Press the Option button again, then Admin Menu (5). Type 22222 for the admin password and press Enter. Then choose Config Server (1), then TFTP Settings (2), then Primary TFTP (1), enter the IP address of your new server, and press Done a half dozen times.

Log back into your server. Switch to the TFTP directory: cd /tftpboot. You'll notice that there are config files for up to 15 phones. Simply choose the extension number you wish to use for each phone and rename the file from 701.cfg to the MAC address of each phone.cfg. Do NOT use hyphens in the MAC address. One final step and you'll be ready to load up your phones. We need to set the correct IP address to tell each phone where your server is located. So... issue the following command using the IP address of your new server instead of 192.168.0.123. Leave the rest of the command as it is!

sed -i 's|192.168.0.0|192.168.0.123|g' /tftpboot/aastra.cfg


Now restart each phone by pressing the Option button and then Restart Phone (6) and then the Restart button. Once the phone reboots, you can make a test call by dialing 1-2-3-4. You can get the latest news by dialing 5-1-1. Or get a weather forecast by airport code (6-1-1) or zip code (Z-I-P).

A Word About Ports. For the techies out there that want to configure remote telephones or link to a server in another town, you'll need to know the ports to remap to your new server from your firewall. Here's a list of the ports available and used by PBX in a Flash. We don't recommend exposing UDP 5038 which is used to communicate with Asterisk via the Asterisk Manager.

TCP 80 - HTTP (needed if you want to access the web sites on your new server from the Internet)
TCP 22 - SSH (needed if you want remote SSH access)
TCP 9001 - WebMin (needed if you want remote WebMin access, not recommended)
UDP 10000-20000 - RTP (needed for SIP communications)
UDP 5004-5037 - SIP (ditto)
UDP 5039-5082 - SIP (ditto)
UDP 4569 - IAX2 (needed for IAX communications typically between Asterisk servers)


Setting Up Trunks for Outgoing and Incoming Calls. If you want to communicate with the rest of the telephones in the world, then you'll need a way to route outbound calls (terminations) to their destination. And you'll need a phone number (DIDs) so that folks can call you. Unlike the Ma Bell world, you need not rely upon the same provider for both. And nothing prevents you from having multiple outbound and incoming trunks to your new PBX. At a minimum, however, you do need one outbound trunk and one inbound phone number unless you're merely planning to talk to other extensions set up on your system. We've actually put all the hooks in place to make it easy for you to interconnect to other Asterisk servers, but we'll save that for another day. For today, we want to get you a functioning system so that you can place outbound calls to anywhere in the world and can receive incoming calls from anywhere in the world. Thanks to our friends at Vitelity, this is not only an easy process, but it's also an incredible deal... but only for PBX in a Flash users.

Vitelity: The Best Provider and Pricing on the Planet. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the PBX in a Flash project gets a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here's a deal you can't (and shouldn't) refuse! Sign up before the end of June, and you can purchase a Tier A DID with unlimited incoming calls for just $3.79 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won't get the special pricing! After the free hour of outbound calling, Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. You can't beat the price OR the call quality! Trust us. We've tried just about everybody. Update: This offer has been extended until July 15.

To sweeten the pot a bit more, we've preconfigured both inbound and outbound Vitelity trunks for you. For the vitel-inbound trunk, all you'll need to do is plug in your username, password, and host assigned by Vitelity and adjust the registration string to match your assigned username and password. In FreePBX, click Setup, Trunks, SIP/vitel-inbound and make the changes. Then adjust the vitel-outbound trunk to reflect your actual username in the fromuser and username entries, your real password in the secret entry, and the correct host provided by Vitelity for your outbound calls, and you're all set. In FreePBX, click Setup, Trunks, SIP/vitel-outbound and make the changes.

To test things out, pick up a phone configured on your system and dial an area code and number of someone in the United States or Canada. Now get someone to call you using your new number. Presto! You have inbound and outbound phone service. And, if you'd like to see just how good Vitelity can be, pick up a phone on your system and dial 678-444-2445. This will connect you to the PBX in a Flash hosted demo applications server at Aretta Communications using Vitelity as the inbound DID provider. Now hang up and place another call to the same server by dialing D-E-M-O which makes a free SIP-to-SIP connection between your new server and our server at Aretta.

An Alternate Outbound Calling Solution. As we said, it costs you almost nothing to add an alternate outbound calling solution to your new system. As luck would have it, adding a second outbound calling provider is now a breeze because AOL just entered the SIP terminations market with a product called AIM Call Out. We wrote about it recently, and you can read the article here. All you need is an AOL or AIM account name and $5 to get you started. The system you've just installed is preconfigured to use AIM Call Out. All you have to do is plug in your username and password, and you can immediately make calls to anywhere in the United States for under 2¢ per minute. Adding international calling is as easy as inserting the correct dial string. If you never use it, it doesn't cost you a dime. So $5 is mighty cheap insurance in our book.

First things first. Sign up for the service at this link. Your username will look something like this: johndoe@aim.com. You also will be assigned a password. Using your web browser, open FreePBX by pointing to the IP address of your new server and choosing Administration, then FreePBX. Type in admin as your username and the password you assigned to your system. From the main FreePBX menu, choose Setup, Trunks, and click on SIP/AIM in the far right column. Scroll down to the Peer Details section of the form and replace yourAIMpassword with your new password. Then replace yourAIMaccountname with your actual AIM account name. Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Setting Up an Alternate DID for Incoming Calls. You also may want to consider a second phone number where people can call you. For example, if Grandma and Grandpa happen to be in another state and still have an old fashioned telephone, you might consider adding an additional DID to your system in their area code. They then can make a local call to reach you by dialing the local DID. On the les.net pay-as-you-go plan, it costs less than a dollar a month plus a penny a minute for the calls. Money well spent if we do say so... and you'll sleep better.

If this setup looks a bit complicated, don't be intimidated. Remember, we're connecting your PBX to the rest of the world so people can call you! With les.net, you have a choice of rate plans for most DIDs. You either can pay $3.99 a month for unlimited inbound calls with two concurrent channels or 99¢ per month and 1.1¢ per minute with four concurrent channels. Just visit their site and click Signup to register. Once you are registered, click Login and then Order DIDs. Pick a phone number. Then click Peers/Trunks and Create New Peer. Write down the Peer Name as you will need it in a minute to set up your connection. Choose SIP for Peer Technology, RFC2833 for DTMF Mode, G.711 for Codecs, Registration for Peer Type, enter the public IP address of your server for Peer Address, make up a secure password and write it down also, specify an Outbound CallerID for your calls, and check the 10-digit dialing box. Leave voicemail unchecked since you'll handle this on your end. Save your changes.

Now choose Your DIDs and click on the one you just ordered. We now need to tie the phone number to the Peer setup you just created above. Click on the DID and select the Route to Peer which you just created. Check the Send DID Prefix box and leave everything else blank. Click Save Changes and you're finished at the les.net end. Now let's set up your inbound DID trunk in Asterisk using FreePBX.

Log into FreePBX using a web browser. Click Setup, Trunks and then Add SIP Trunk. Fill in the CallerID and then drop down to the Outgoing Settings section of the form. For Trunk Name, use the Peer Name that you created above and wrote down. It ought to look something like this: 1092832198. For Peer Details, enter the following using the Peer Name and Password you assigned at les.net:

canreinvite=no
context=from-trunk
fromuser=1092832198
host=did.voip.les.net
insecure=very
nat=yes
secret=yourpassword
type=peer
username=1092832198

For Incoming Settings, use from-pstn for the User Context and enter the following User Details:

canreinvite=no
context=from-pstn
dtmfmode=rfc2833
insecure=very
nat=yes
type=user

For the registration string, enter a string like the following using your Peer Name and Password:

1092832198:yourpassword@did.voip.les.net/1092832198

Now click the Submit Changes button and then Apply Configuration Changes and Continue with Reload.

Choosing a Preferred Provider. Finally, you'll need to decide whether to use AOL or Vitelity as your primary terminations provider. HINT: Vitelity is less costly. So we've set them up as your primary terminations provider with AOL as the backup. This is handled in FreePBX in the Outbound Routes tab under the AllCalls entry.

A Word About Mondo Rescue. We would be remiss if we didn't mention what a fantastic open source product Mondo Rescue is. It's the sole reason that today's build was possible. Our special thanks go to the development team: Bruno Cornec, Andree Leidenfrost, and Hugo Rabson. It is the first (and only) backup software for Linux builds that actually works reliably. The best way to prove that for yourself is to download this build and try it for yourself on your Everex gPC2. It has much more flexibility than what you will experience, but that would take another dozen pages to explain. We'll save that for another day. In the meantime, if you'd like more information, visit the Mondo Rescue web site.

Where To Go From Here. Well, we've covered a good bit of territory today so we're going to save the really fun stuff for our next installment. In the meantime, you have a new phone system that works. And there are a number of PDF documents in the /root folder on your new system which are worth a read. Better yet, you can browse through all of the documentation which is available for PBX in a Flash by going here. You also can dial D-E-M-O on your new system and see just how powerful direct SIP connections can be to other Asterisk hosts (in this case, ours!)... at no cost. Finally, you can log into your server and type help-pbx for access to a treasure trove of additional features. Enjoy!

Continue reading Part II...

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:

#!/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. Just dial 678-444-2445 from any touchtone phone.

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.

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 infr