Category Archives: Telephony

Just when you thought the National Do-Not-Call Registry was getting you some peace and quiet during the dinner hour, VoIP telephony comes along to give the telemarketers a brand new universe to pollute. And, of course, the politicians exempted themselves and non-profits from the Do-Not-Call rules anyway. Thanks to Katrina and local elections in November, you can expect a wave of unwanted dinnertime calls from your best friends at campaign headquarters or the Fraternal Order of Police. Lucky for you, there's an Asterisk® PBX standing between the telemarketers and your dinner table. Here are a few simple additions you can make to your Asterisk PBX setup to all but eliminate unwanted callers from your life. There are three types of protection we'll address. First, you can build a separate context to handle callers without CallerID. Second, you can send a special information tone to certain callers to block autodialers. And finally, when all else fails, you can quickly place certain numbers in a BlackList database to make sure it's the last time that folks using that number ever disturb you again.

Managing CallerID-less Callers. Not all callers without a CallerID name and number are bad people, at least not quite. So we want to structure our treatment of calls without CallerID in such a way that we don't discard a call that might be important. There are a couple of things you can do to manage these calls. First, you can have Asterisk prompt such callers to either say their name or to key in their phone number. Our preference is recording the name of the caller because hearing the caller speak gives you a good idea whether you want to take the call whereas asking a caller to enter their phone number does nothing to deter really obnoxious telemarketers. With either of these options, our approach (which we previously covered in our security column) is to prompt the caller for the information, park the caller with music on hold, and then announce the call and play back either the caller's name or number. You then have the option of picking up the parked call or leaving the caller parked until they're automatically disconnected.

Our other recommendation for calls without CallerID is to send a special information tone when the call is answered. For those of you that spent $40 on a Telezapper, we're sorry. Asterisk can do it for free. And it really does work with many autodialers used by telemarketers. In fact, with most such systems, once the autodialer receives the special information tone, it places your number in their do-not-call database so you'll never be bothered again. Here's the code we previously recommended to handle calls without CallerID. First, for Asterisk@Home users and others using the Asterisk Management Portal, you tell Asterisk to send incoming calls to your AutoAttendant context. Of all the Asterisk@Home problems we read about, the number 1 issue hands down is incoming calls either ringing with a fast busy or being dropped immediately into voicemail. You fix both problems by deleting the current contents of your [from-sip-external] context and adding the following GoTo command to the [from-sip-external] context in the extensions.conf file. This will send incoming callers to your AutoAttendant (shown below).

exten => _.,1,Wait(1)
exten => _.,2,Goto(from-external-custom,s,1)

And then you drop the following AutoAttendant context into the bottom of your extensions_custom.conf config file. As we've mentioned before, if you cut-and-paste the code below, you'll need to manually replace the typographic quotation marks with regular quote marks, or Asterisk gets sent into the ozone.

[from-external-custom]
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,Wait(1)
exten => s,3,SetMusicOnHold(default)
exten => s,4,GotoIf($["${CALLERIDNUM}" = ""]?who-r-u,s,1)
exten => s,5,GotoIf($["foo${CALLERIDNUM}" = "foo"]?who-r-u,s,1)
exten => s,6,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?who-r-u,s,1)
exten => s,7,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?who-r-u,s,1)
exten => s,8,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => s,9,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)
exten => s,10,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?who-r-u,s,1)
exten => s,11,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => s,12,DigitTimeout,3
exten => s,13,ResponseTimeout,3
exten => s,14,Background(custom/welcome)

exten => i,1,Playback(wrong-try-again-smarty)
exten => i,2,Goto(s,16)


And finally you add the following two contexts to the bottom of the extensions_custom.conf file to handle the unidentified callers. The extension to ring to announce unidentified callers (204 in this example) is in line 70,5 below.

[who-r-u]
exten => s,1,Background(privacy-unident)
exten => s,2,Background(vm-rec-name)
exten => s,3,Wait(2)
exten => s,4,Record(/tmp/asterisk-stranger:gsm|5|15)
exten => s,5,Background(pls-hold-while-try)
exten => s,6,Goto(ext-park,70,1)
exten => s,7,VoiceMail(204@default)
exten => s,8,Playback(Goodbye)
exten => s,9,Hangup

[ext-park]
exten => 70,1,Answer
exten => 70,2,SetMusicOnHold(default)
exten => 70,3,SetCIDNum(200|a)
exten => 70,4,SetCIDName(Parked Call Info|a)
exten => 70,5,ParkAndAnnounce(silence/9:asterisk-friend:/tmp/asterisk-stranger:vm-isonphone:at-following-number:PARKED|40|local/204@from-internal|who-r-u,s,7)
exten => 70,6,Hangup

A footnote to all of this technology is that we personally receive so few legitimate calls from callers without CallerID that we've modified the [who-r-us] context to simply send all these callers straight to voicemail. We'll get a phone alert and an email whenever a new voicemail arrives so, if it's some sort of emergency, we can respond by returning the call immediately. Haven't seen one yet!

BlackListing. And then there are the smart telemarketers, and we'd put the Baby Bells at the top of this list. These are organizations that intentionally provide a fictitious CallerID number just to get around systems that block calls with no CallerID. They're still selling something, and they're just as annoying. They slip into your home under an exception to the Do-Not-Call Registry for "calls from organizations with which you have established a business relationship." In other words, if you buy local phone or cable TV service, these folks have a blank check to annoy the hell out of you ... forever! That's their interpretation of the statute anyway.

As luck would have it, Asterisk@Home 1.5 handles blacklisting callers using its internal database (ast_db) so you never have to take another annoying call from them. Just pick up your phone after an unwanted call, and press *32. That's it. Not much in what follows is original by the way. Our special thanks to Jacken's Blog for documenting all of this. All we've done is revise their code a bit to make it fit the configuration laid out in our other Asterisk@Home tutorials. Note also that problems have been reported using this code with the Asterisk@Home 2.0 betas, but we'll address that down the road as well. To implement BlackListing, we're going to add a line at the top and bottom of our AutoAttendant code and then renumber the 's' extension commands. We also need to adjust the pointer on line i,2 to goto s,17. So the new code looks like this. The way the LookupBlacklist command works is that, if a CallerID number is found in the BlackList database, execution jumps to line s,104 (3 + 101). From there, we send the call to a "special" context to handle blacklisted callers.

[from-external-custom]
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,Wait(1)
exten => s,3,LookupBlacklist ; If CID blacklisted, goto 104
exten => s,4,SetMusicOnHold(default)
exten => s,5,GotoIf($["${CALLERIDNUM}" = ""]?who-r-u,s,1)
exten => s,6,GotoIf($["foo${CALLERIDNUM}" = "foo"]?who-r-u,s,1)
exten => s,7,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?who-r-u,s,1)
exten => s,8,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?who-r-u,s,1)
exten => s,9,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => s,10,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)
exten => s,11,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?who-r-u,s,1)
exten => s,12,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => s,13,DigitTimeout,3
exten => s,14,ResponseTimeout,3
exten => s,15,Background(custom/welcome)
exten => s,104,Goto(custom-blacklisted,s,1)

exten => i,1,Playback(wrong-try-again-smarty)
exten => i,2,Goto(s,17)

Now we need to drop in four BlackList contexts to let you respond to BlackListed callers and to manage your BlackList process using any touchtone phone. So, at the bottom of the extensions_custom.conf file, add the following:

[custom-blacklist-last]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,DBget(number=CALLTRACE/${CALLERIDNUM}) ; goto 104 if no lastcaller
exten => s,4,GotoIf($"${number}" = ""?104) ; also if it's blank (caller id blocked)
exten => s,5,Playback(privacy-to-blacklist-last-caller)
exten => s,6,Playback(telephone-number)
exten => s,7,SayDigits(${number})
exten => s,8,Wait,1
exten => s,9,Background(press-1)
exten => s,10,Background(or)
exten => s,11,Background(press-star-cancel)
exten => s,12,Hangup
exten => s,104,Playback(unidentified-no-callback)
exten => s,105,Background(goodbye)
exten => s,106,Hangup
exten => 1,1,DBput(blacklist/${number}=1)
exten => 1,2,Playback(privacy-blacklisted)
exten => 1,3,Wait,1
exten => 1,4,Background(goodbye)
exten => 1,5,Hangup
exten => t,1,Background(goodbye)
exten => t,2,Hangup
exten => i,1,Background(goodbye)
exten => i,2,Hangup
exten => o,1,Background(goodbye)
exten => o,2,Hangup

[custom-blacklisted]
exten=>s,1,Answer
exten=>s,2,Wait(1)
;exten=>s,3,Playback(nbdy-avail-to-take-call)
;exten=>s,4,Playback(carried-away-by-monkeys)
;exten=>s,5,Playback(lots-o-monkeys)
exten=>s,3,Background(tt-allbusy)
exten=>s,4,SetMusicOnHold,default
exten=>s,5,WaitMusicOnHold,30
exten=>s,6,Background(thank-you-for-calling)
exten=>s,7,Background(goodbye)
exten=>s,8,Congestion
exten=>s,9,Hangup

The [custom-blacklist-last] chunk of code automatically adds the phone number of your last incoming call to your BlackList. The [custom-blacklist-add] context lets you manually add a number to your BlackList. The [custom-blacklist-remove] context lets you remove a number that's already in your BlackList. And the [custom-blacklisted] context actually processes incoming callers who are on your BlackList. As currently written, the caller will get a message that "all members of the household are currently assisting other telemarketers ..." followed by music on hold. After 30 seconds, they are kissed goodbye with a congestion tone. I've also commented out the Jacken's Blog approach which is equally annoying. So take your pick.

The only remaining step to get all this working is to designate some extensions that will be dialed to access the three custom BlackList management contexts above. These need to be placed within the [from-internal-custom] context of your extensions_custom.conf file. Feel free to make up your own extension numbers so long as they don't conflict with existing extensions on your system. And be sure to change the Authenticate password in each of the three lines below. Once you add the extensions, reload Asterisk and BlackList someone you love... or at least someone you used to love.

exten => *30,1,Authenticate(45678)
exten => *30,2,Goto(custom-blacklist-add,s,1)

exten => *33,1,Goto(custom-blacklisted,s,1)


How To Review Your BlackList. One final piece remains for our puzzle today. At some point down the line, you may want to review every number that's been entered into your BlackList. Here's how. Using SSH or Putty, connect to your Asterisk server and log in as root. Start up the Asterisk Command Line Interface (CLI) with the command asterisk -r. Now enter the following command at the asterisk*CLI> prompt: database show blacklist. You can manually delete an entry while you're here with the command: database del blacklist 0123456789 1. Don't forget the trailing 1. To manually add an entry to the database, enter the command: database put blacklist 0123456789 1.

You're an expert now. So just sit back and wait for the Bad Guys to call. They will.

Some Recent Nerd Vittles Articles of Interest...

If you haven't noticed, useful Caller ID (meaning a number and a name display) is pretty much a bust in the VoIP marketplace except for calls originating from Baby Bell-controlled local phone numbers. And, with some VoIP providers, getting a CallerID name with any incoming call is a rarity. Jeff Pulver has proposed a new national database where you can list yourself. In fact, you can sign up today. But, suffice it to say, it isn't soup just yet. Known in the trade as CNAM service, many telephony service providers simply throw incoming names in the bit bucket unless you are one of their subscribers. The Baby Bells are among the most notorious. Some don't even provide CNAM service from other areas of the country unless the caller is part of the local carrier's feifdom. And I guess if I charged $40 for basic local phone service with CallerID, I'd want to keep my monopoly, too. We'll have more on the pricing issue at the end of today's article.

2008 Update. For the latest in CallerID name lookup software written in PHP, visit our Best of Nerd Vittles site. For PBX in a Flash users that want a Perl version, check out the PBX in a Flash Forum.

For those of you wrestling with Caller ID on your Asterisk® PBX, we have three solutions today and more to come. Today's perl AGI utility was developed initially by Tom Vile at Baldwin Technology Solutions. Tom has graciously agreed to let us share the code with you. Thanks, Tom! It lets you intercept incoming calls to your Asterisk box and pass the CallerID number to AT&T's AnyWho.com for a reverse number lookup to decipher the CallerID name. Whether this comports with the AnyWho terms of service, we'll leave for you to resolve. Suffice it to say, the "phone company" has always maintained that the phone book information is copyrightable. And the Supreme Court of the U.S. has held just the opposite. This is not legal advice, just some historical background for you to digest before proceeding.

Once we started looking at Tom's code, we decided it might be a good time to learn Perl so you've been forewarned that nothing in the solutions which follow will qualify as elegant coding other than Tom's original handiwork, of course. But the stuff does work. What we've added to Tom's original code are two enhancements. First, you can opt to use Google for reverse number lookup if you'd like. And second, you can tie reverse number lookups into the AsteriDex web-based MySQL database application which we previously built and which you can download and use for free here. Particularly for home and small business use, the universe of incoming callers is fairly small so you may find that AsteriDex is the best solution. This is particularly true if many of your incoming calls are from cell phones since few of the American carriers associate real names with their CallerID numbers. Some do provide the city of the caller in the CallerID name. Others refer to us all as Wireless Caller(s). Gee, what a hint. Consequently, neither AnyWho nor Google have many cell phone numbers in their databases. Call it a feature. The bottom line is you can mix and match AnyWho, Google, and AsteriDex lookups as you see fit by simply setting "on" and "off" flags for each of the three services.

Footnote: As of November 13, we've added another lookup function for FoneFinder.net. This one's a little different in that it returns the city and provider type for phone numbers matching the area code and first three digits of the caller's number. It also has the lowest precedence and can be activated to at least return the city name and provider type for callers where no other information is available from the other services.

Overview. The way this works is that incoming calls will be processed through an AGI script that you configure with your lookup preferences. We recommend you use this script with Asterisk@Home because it comes bundled with all the MySQL, Apache, PHP, and Perl stuff you'll need to make everything work. The script as received does nothing since all three lookups are disabled. That lets you choose which services to activate and conveniently moves the legal monkey from our back to yours. Didn't go to law school for nothing, did we? Assuming you turn on all three lookups, the AnyWho lookup is processed first. If a match is found, the Caller ID name is added to the existing Caller ID number replacing whatever name entry already was picked up for the incoming call. If no match is found, the existing CallerID number and name are left as they were received for the incoming call. The CallerID number is then passed to the Google phonebook where the process is repeated. If there's a match, the CallerID name is replaced with the name found in the Google search. If not, nothing changes. Finally, if you're using AsteriDex as your personal phone book, the CallerID number will be looked up in your AsteriDex database. If there's a match, whatever you entered as the name for the first matching phone number entry will be picked up as the CallerID name for the call ... so these names can be as obnoxious as you choose to make them. Note that the AsteriDex lookup is a crude search. If you've entered the same phone number for three different people in the same house, then only the first one it finds will be used. You'll know which one it is when you receive the first call from this number. So, the bottom line is this: AsteriDex lookups take precedence over Google and AnyWho lookups, and Google lookups take precedence over AnyWho lookups. And, at least for the short term, if you want any meaningful information about cell phone callers and most VoIP callers, you'll need to put their names and numbers in your AsteriDex database.

To get started, download the calleridname.agi script from here. Then copy it to the /var/lib/asterisk/agi-bin folder on your Asterisk server. Log in to your Asterisk server as root and change the ownership of the file: chown asterisk:asterisk /var/lib/asterisk/agi-bin/calleridname.agi. And change the permissions: chmod 775 /var/lib/asterisk/agi-bin/calleridname.agi. To activate the services you want to use, edit the calleridname.agi script: nano -w /var/lib/asterisk/agi-bin/calleridname.agi. CAUTION: Before you open the file with nano, be sure your editing window is at least 180 characters wide unless you use the -w switch, or some of the commands in the file will be truncated. Then nothing works! Nano doesn't do word wrap in a Perl-friendly way if left to its own devices. Once you open the file, beginning on line 10, you'll see the following entries:

$Fonetastic = '0' ;
$AnyWho = '0' ;
$Google = '0' ;
$Asteridex = '0' ;

For each service you want to activate, change the '0' to '1' and then save the file: Ctrl-X, Y, then press Enter key. Now we're ready to reconfigure your incoming call dialplan.

If you've been following along with our other tutorials, you should already have the Stealth AutoAttendant in place to handle your incoming calls. If not, start there ... or you're on your own. After making the security modifications, here's how our autoattendant code looks in the extensions_custom.conf config file:

[from-external-custom]
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,Wait(1)
exten => s,3,SetMusicOnHold(default)
exten => s,4,GotoIf($["${CALLERIDNUM}" = ""]?who-r-u,s,1)
exten => s,5,GotoIf($["foo${CALLERIDNUM}" = "foo"]?who-r-u,s,1)
exten => s,6,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?who-r-u,s,1)
exten => s,7,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?who-r-u,s,1)
exten => s,8,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => s,9,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)
exten => s,10,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?who-r-u,s,1)
exten => s,11,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => s,12,DigitTimeout,3
exten => s,13,ResponseTimeout,3
exten => s,14,Background(custom/welcome)

We're going to introduce another new trick in the incoming dial plan. What we want to do is answer the call, do some processing, and then pass the call to where it's supposed to go. The trick is that we don't want the caller to know we've already answered the call while we're doing the processing. So what we're going to do is play a fake ringing tone to the caller so the caller doesn't get bored. Just insert a new third line in the dialplan that looks like this: exten => s,3,Playtones(ring) and then renumber the remaining lines. Next we want to add our new CallerIDName lookup immediately before the custom/welcome message plays: exten => s,17,AGI(calleridname.agi) and then renumber the lines. When you're all finished, your code should look like this:

[from-external-custom]
exten => s,1,Zapateller(answer|nocallerid)
exten => s,2,Wait(1)
exten => s,3,Playtones(ring)
exten => s,4,SetMusicOnHold(default)
exten => s,5,GotoIf($["${CALLERIDNUM}" = ""]?who-r-u,s,1)
exten => s,6,GotoIf($["foo${CALLERIDNUM}" = "foo"]?who-r-u,s,1)
exten => s,7,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?who-r-u,s,1)
exten => s,8,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?who-r-u,s,1)
exten => s,9,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?who-r-u,s,1)
exten => s,10,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?who-r-u,s,1)
exten => s,11,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?who-r-u,s,1)
exten => s,12,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?who-r-u,s,1)
exten => s,13,DigitTimeout,3
exten => s,14,ResponseTimeout,3
exten => s,15,AGI(calleridname.agi)
exten => s,16,Background(custom/welcome)

Remember that, if you cut-and-paste the code above, you'll need to manually fix the typographic quotation marks to make them regular quote marks that Asterisk can understand or disaster awaits. Once you save your changes and reload Asterisk, you should be good to go. Start up the Asterisk Command Line Interface (CLI) and make a test call to yourself. You should see something like this in the CLI display:

calleridname.agi: CALLERID IS: 3035551616 <3035551616>
calleridname.agi: Checking 303 555 1616...
calleridname.agi: Fonetastic.US lookup disabled.
calleridname.agi: AnyWho lookup disabled.
calleridname.agi: Ready for Google lookup...
calleridname.agi: Google match. New CallerIDName = R. Smith
calleridname.agi: Ready for AsteriDex lookup...
calleridname.agi: AsteriDex match. New CallerIDName = Robbie the Nerd


Money-Saver Tip of the Week. We regularly hammer BellSouth and the other RBOCs for their pricing policies on home phone service so it's great to finally have something nice to say about our hometown company. Actually, BellSouth might not think this is too nice, but we sure do. A post on DSL Reports this week had this helpful tip for those with BellSouth DSL service that would prefer not to keep paying $40 a month for a BellSouth phone line you never use. You can suspend your phone service for up to six months and reduce your monthly line costs by 50% or more while still retaining your $24.95 DSL service through BellSouth. You can even get a recorded message referring callers to your new [VoIP] number at no charge. For details, visit this BellSouth link. To put things in proper perspective, this means you can suspend your BellSouth line, order two new TelaSIP VoIP lines with unlimited U.S. long distance on both lines for $14.95 a month, notify your BellSouth callers of your new number, and still put $5 in the bank each month compared to what you're paying BellSouth today for just one line with pay-through-the-nose long distance access. Now that's a sweet deal! For those that are curious, a garden-variety residential line from BellSouth in Atlanta with nothing other than CallerID and dial tone runs $39.75 per month with tip and taxes, and I've got this month's statement to prove it. Does it take a collision with a freight train for the RBOCs to wake up before all their residential customers have jumped ship just like their pay phone customers did? Probably so.

Sony Anyone? Just Say No! If you missed the latest attack on your home computer this week, don't worry. It wasn't a malicious virus creator this time. It's S-O-N-Y. Before you spend another dime with Sony, read this C|NET article. Here's a brief snippet:

"You buy a CD. You put the CD into your PC in order to enjoy your music. Sony grabs this opportunity to sneak into your house like a virus and set up camp, and it leaves the backdoor open so that Sony or any other enterprising intruder can follow and have the run of the place. If you try to kick Sony out, it trashes the place. And what does this software do once it's on your PC ..."

Other Asterisk Tutorials. There are numerous additional articles in this Asterisk HOW-TO series to keep you busy. You can read all of them by clicking here and scrolling down the page. We recommend reading at least the first four or five articles from the bottom up so that the learning curve is less painful. Then you can skip around to your heart's content.

This is the third and final article in our series showing how to deploy a Bluetooth Proximity Detection system with your Asterisk@Home PBX. Parts I and II are here. Today we finish up and activate everything so that your Asterisk® PBX system will track your whereabouts and automatically transfer incoming calls in your home or office to your cellphone or any other phone whenever you leave home base carrying your bluetooth-enabled cellphone or your bluetooth headset. And, when you return, calls automatically will begin flowing to your local extensions just as they did before you left your home or office. For our Mac users, we'll have a follow-up article some day soon showing you how to tie proximity detection into the home automation server we previously built with Indigo. Then, as you arrive home in the evening, you can turn on your lights, and stereo, and hot tub just by pulling up in front of your house with your Bluetooth headset or cellphone.

NOTE: This article has been updated to take advantage of TrixBox, freePBX, and the iPhone. For the current article, click here.

Last week we showed you how to use your new Asterisk@Home 2 PBX machine to detect the presence or absence of your bluetooth phone or headset. And we used freely available open source tools for Linux to do the tracking. Our game plan for today is to show you how to relay that information from your Bluetooth Proximity Detection system to your main Asterisk system in real time. And then we'll modify the dialplan on your main Asterisk system a bit so that it intelligently routes incoming calls after first deciphering whether you are IN or OUT of your home or office. For purposes of this tutorial, we're assuming that you have taken our advice and implemented a second Asterisk@Home server (the version 2 beta) to take advantage of the bundled CentOS/4 Linux operating system. Our design also assumes that both your main Asterisk machine and your new Asterisk proximity detection system are both behind the same firewall on the same internal network. So we'll begin today by finishing the detection tasks on the new Asterisk server, and then we'll turn our attention to your primary Asterisk server, i.e. the system that takes and makes your phone calls. Update: Now that Asterisk@Home 2.1 is shipping, you may decide you'd prefer to deploy this entire system on a single server. If so, print this article and then go here for the instructions needed for a single server deployment.

Overview of Bluetooth Proximity Detection Design. If you've been following along with the two previous articles in this series, then you already have your Asterisk@Home 2 PBX server running with full Bluetooth support. If not, start there. The real trick to making Bluetooth Proximity Detection work for this project was using hcitool name 00:03:89:43:84:e2 with the actual MAC address of your Bluetooth headset or phone. This utility returns a null string if the Bluetooth device is not found and returns the name of the device if it is found. We'll take advantage of that information to drive our proximity detection system by simply writing the results of this test to a text file once a minute. We then can use a Linux bash script to execute a certain action based upon whether your Bluetooth device is IN range or OUT of range. As usual, we've chosen the easy way. So our plan is to copy a null file to your primary Asterisk server when you're OUT, and we'll copy a one-byte file to your primary Asterisk server when you're IN. Then we can run a similar script on the primary Asterisk server to check whether you're IN or OUT each time a phone call arrives. We'll only upload the null or one-byte file when the status changes to keep the processing load low. If you want to track multiple people with multiple Bluetooth devices, then rename the ruhome file to include the name of the person that's carrying the device, e.g. wardruhome. Then it'll be easy to implement this for multiple people. Just repeat the steps.

Configuring Files for Asterisk 2 Server. The files we'll be using in our Proximity Detection System can be downloaded here and unzipped into the proximity folder on your Desktop. Don't modify null.file or notnull.file because doing so will probably change their file sizes, and then none of this works. We are depending upon the notnull.file remaining a zero byte file! You do need to edit the ruhome file whether you rename it or not. Call it up with a text editor and modify the following settings to match your configuration:

mainasteriskbox=192.168.0.104
deviceuser=WARD
devicemac=00:03:89:43:84:e2

We recommend upper case letters for the deviceuser just so it will be easy to identify. The devicemac is the MAC address of your Bluetooth device to be monitored. And mainasteriskbox is the private IP address of your main Asterisk server. Once you make these changes, save the file in text-only format and then copy the following three files in the proximity folder to the /root directory on your Asterisk2 Proximity Detection Server: ruhome, null.file, and notnull.file. Then log in to your Asterisk2 server as root, and move to the /root folder: cd /root. Now adjust the file permissions on script you put there substituting the name you gave your script: chmod 775 ruhome.

If you read through the ruhome script, you'll notice that it copies either null.file or notnull.file to your primary Asterisk server whenever there's a change in the presence of your bluetooth device, and it names the file to match the assigned deviceuser. To make the copy process work, we need to use the scp utility that is bundled with Linux. There's only one wrinkle. SCP prompts for a password whenever you attempt to log in to your other Asterisk box, and that would preclude our being able to handle this as a background job with no user interaction. Luckily, there's a simple solution: a public/private key pair. If you want the complete tutorial, go here. Otherwise, just do the following. Log in to your new Asterisk@Home 2 server as root. From the command prompt, issue the following command: ssh-keygen -t rsa. Press the enter key three times. You should see something similar to the following. The file name and location in bold below is the information we need:

Generating public/private rsa key pair.
Enter file in which to save the key (/root/.ssh/id_rsa):
Enter passphrase (empty for no passphrase):
Enter same passphrase again:
Your identification has been saved in /root/.ssh/id_rsa.
Your public key has been saved in /root/.ssh/id_rsa.pub.
The key fingerprint is:
1d:3c:14:23:d8:7b:57:d2:cd:18:70:80:0f:9b:b5:92 root@asterisk1.local

Now copy the file in bold above to your main Asterisk server from your secondary Asterisk2 server using SCP. The command should look like the following (except use the private IP address of your primary Asterisk server instead of 192.168.0.104). Provide the root password to your primary Asterisk server when prompted to do so.

scp /root/.ssh/id_rsa.pub root@192.168.0.104:/root/.ssh/authorized_keys

Once the file has been copied, let's try to log in to your primary Asterisk server from your Asterisk 2 Proximity Server with the following command using the correct IP address, of course:

ssh root@192.168.0.104

You should be admitted without entering a password. If not, repeat the drill or read the complete article and find where you made a mistake. Now log out of the primary server by typing exit.

The final step on the Asterisk2 Proximity Server is to schedule this script to run once a minute to keep track of your comings and goings. While still logged in as root, issue the following commands:

export EDITOR=nano
crontab -e

This will start up the Crontab utility using our favorite editor, Nano. Now enter the command below substituting the correct name of your script:

* * * * * /root/ruhome > /dev/null

If you'd prefer to only run the script every minute between 6 in the morning and 9 at night, the code would look like what's shown below. Just choose the hours desired and remember that your IN and OUT status will be frozen at the exact moment the script completes its run at 8:59 p.m. if you use the settings below. And it won't check your bluetooth proximity again until 6:00 a.m. tomorrow morning.

* 6-21 * * * /root/ruhome > /dev/null

Once you've gotten the entry the way you want it using the actual name of the script you placed in the root folder, save your crontab entry: Ctrl-X, Y, and press Enter key. Now turn on your Bluetooth device and move it within range of your Asterisk2 server, count to 60, and then check the /tmp folder on your primary Asterisk server for a file with the name you specified for deviceuser above, e.g. ls -all /tmp/WARD. The size of the file should be 1 indicating that your specified Bluetooth device is within range of your Asterisk2 server. Now turn the Bluetooth device off and wait another minute or two. Repeat the steps above and make sure the file size has shrunk to 0. When all this works according to plan, you're ready to move on and configure your primary Asterisk server to manage your incoming calls based upon whether you're IN or OUT.

Configuring Your Main Asterisk Server for Proximity Detection. As you learned above, the Asterisk2 proximity detection server will actually do the heavy lifting by figuring out when you're IN and when you're OUT. It then will upload the appropriate file to your primary Asterisk server when your status changes. But we'll still need a little script on your primary Asterisk server to actually check your status whenever an incoming call arrives. That's the job of the homecheck.agi script in the proximity folder you downloaded. Before it will work, the deviceuser specified at the top of this file must match the name you assigned to deviceuser in the ruhome script. So edit the homecheck.agi script and then save the file as plain text. Then copy it to /var/lib/asterisk/agi-bin on your primary Asterisk server. Log in to that server as root and change the ownership of this file: chown asterisk:asterisk /var/lib/asterisk/agi-bin/homecheck.agi. And then set the permissions: chmod 775 /var/lib/asterisk/agi-bin/homecheck.agi.

The remaining tasks on the primary Asterisk server we'll handle using the Asterisk Management Portal (AMP) so, using your favorite web browser, connect to the primary Asterisk server and then choose AMP->Maintenance->Config Edit and click on extensions_custom.conf. If you're using our Stealth Autoattendant, then locate the code in your extensions_custom.conf file. All that we need to modify is the timeout section of the code: exten=>t. So you'll still be able to take advantage of your AutoAttendant even when you're away from your home or office. Here's what we are using:

exten => t,1,Background(pls-hold-while-try)
exten => t,2,AGI(homecheck.agi)
exten => t,3,GotoIf($["${WARD:0:2}" = "OU"]?custom-outhouse,s,1:4)
exten => t,4,Dial(local/222@from-internal,20,m) ; Ring group 222 dials all phones
exten => t,5,VoiceMail(204@default) ; Default voicemail account for your home
exten => t,6,Hangup

NOTE: You can't cut and paste the above code unless you manually change the typographic quotation marks in the third line to normal quotes that Asterisk understands. There are two sets of them! Also change WARD to the deviceuser name you assigned above in your script files. The final piece of code you'll need is the context to actually process the incoming calls when you are out of the house or office, custom-outhouse. Just cut-and-paste it to the bottom of extensions_custom.conf. As usual, you can customize it to your heart's content, but here are a couple ideas from our own system. Once you're finished, save your changes and then reload Asterisk.

[custom-outhouse]
;exten => s,1,Dial(SIP/6781234567@telasip-gw,60,m) ; This approach just quietly sends callers to your cell phone using the telasip-gw trunk
;exten => s,2,VoiceMail(204@default)
;exten => s,3,Hangup

exten => s,1,Wait(1) ; This approach plays a custom message which we previously showed you how to do. Here's the message:
exten => s,2,Background(custom/nothome) ; No one's home. To try our cellphones, press 1 for Mary or 2 for Ward. Or press 3 to leave a message.
exten => s,3,DigitTimeout,4
exten => s,4,ResponseTimeout,5
exten => t,1,VoiceMail(204@default)
exten => t,2,Hangup
exten => i,1,Playback(wrong-try-again-smarty)
exten => i,2,Goto(s,1)
exten => o,1,VoiceMail(204@default) ; if the caller presses 0, then it's off to voicemail they go
exten => o,2,Hangup
exten => h,1,Hangup
exten => 1,1,Background(pls-hold-while-try)
exten => 1,2,Dial(SIP/6781234567@telasip-gw,60,m)
exten => 1,3,VoiceMail(204@default)
exten => 1,4,Hangup
exten => 2,1,Background(pls-hold-while-try)
exten => 2,2,Dial(SIP/6782345678@telasip-gw,60,m)
exten => 2,3,VoiceMail(204@default)
exten => 2,4,Hangup
exten => 3,1,VoiceMail(204@default)
exten => 3,2,Hangup

Well, that should do it. Once you restart your Asterisk server, you can place a call to your home while you're IN and while you're OUT carrying your Bluetooth device to make sure things are doing what they should. Now all you have to do is to remember to recharge your Bluetooth device each night. Enjoy!

There's now a fourth article in this series which you can read by clicking here.

Coming Attractions. No, we haven't forgotten. We still have our incoming fax server to build, and we want to start using Asterisk's built-in Blacklist database tools. Then we'll be turning our attention to using SSH and SCP to build automated backups of your primary Asterisk server using the password trick we learned in the proximity detection article today. And now that you have a second Asterisk server, you might as well learn how to connect them together so that you can make calls to extensions on either server and make outgoing calls using either server. So stay tuned!

Other Asterisk Tutorials. There are numerous additional articles in this Asterisk HOW-TO series to keep you busy. You can read all of them by clicking here and scrolling down the page. We recommend reading at least the first four or five articles from the bottom up so that the learning curve is less painful. Then you can skip around to your heart's content.

Say It Ain't So. Sony reportedly is taking the low road and employing a rootkit to hide copy-protection mechanisms on its latest CDs and your PC, just what every virus creator on the planet has yearned for. Before you buy anything else from Sony/BMG, read this. Update: A recent article indicates that Sony's rootkit may, in fact, "phone home" to Sony with details of your music listening habits as well as your IP address. And what does Sony have to say about the controversy? They apparently don't think most folks are smart enough to know what a rootkit is. And what does Sony think you should be able to do with your newly purchased CD? Read this eye-opener before you hand over any more money to Sony. Let the lawsuits and prosecutions begin.

This is the second in our series of articles showing how to deploy a Bluetooth Proximity Detection system with Asterisk®@Home. Part I is here. When we’re finished, your system will automatically transfer incoming calls in your home or office to your cellphone or any other phone whenever you leave home base carrying your bluetooth-enabled cellphone or your bluetooth headset. You’ll recall that we recommended the headset approach because cellphones have a nasty habit of putting themselves and their bluetooth adapter to sleep when you’re not on the phone. If bluetooth on the phone is sleeping, we lose our ability to detect your comings and goings so be reasonable and do it our way. Use a bluetooth headset. Once you remove the earpiece, the bluetooth headset fits comfortably in your pocket and isn’t much larger than a flash drive. For our purposes the bluetooth headset will be functioning primarily as an electronic key although there’s no reason you can’t also use it in conjunction with either your bluetooth cellphone, or a softphone connected to your primary Asterisk@Home PBX, or all of the above. The major difference in our approach and some of the other proximity detection systems which (still) are on the drawing boards is cost. Our bluetooth headset "key" costs roughly $30 delivered to your door. Most of the corporate dream systems require a $200 badge (to do the same thing) and then an incredibly expensive server (to do what we’re doing with an old clunker PC). So, yes, open source technology is a very good thing for all of us. And it deserves your financial support. Here’s a link if you’d like to make a contribution in any amount to the Asterisk@Home project. End of sermon.

NOTE: This article has been updated. For the current article, click here.

Installing Asterisk@Home. Today, we need to do a lot of the grunt work to get our system configured with the necessary software to support proximity detection. You’ll recall from the previous article that we’ve decided to use the Asterisk@Home 2 beta build for this project because we need Linux CentOS/4 that’s included to make everything work. So Step 1 is to download and install the Asterisk@Home 2 beta. From our previous Asterisk@Home installation tutorial, you’ll recall the drill involves downloading the .iso image, burning a CD, finding a clunker PC with a hard disk you don’t mind wiping clean (a WalMart special (inset) oughta work just fine!), booting from the install CD after connecting the new PC to your network behind the same firewall as your primary Asterisk@Home system, and then answering a few prompts. If you need a refresher course, review the original installation article or visit the new Asterisk@Home Wiki. Once you get Asterisk@Home 2 up and running, you need to secure your system as outlined in the above article. And then review the security article that will tell you how to lock down MySQL. Now run yum -y update to get CentOS/4 current, and you’re all set. Finally, a word about the beta itself. Some folks get nervous about using beta software. Don’t. Remember, we’re only using Asterisk@Home 2 for proximity detection at this juncture, and nothing in CentOS/4 is beta. By the time we get everything working, the full-blown Asterisk@Home 2 product will probably be on the street, and you can repeat this drill in your sleep to get ready for our next mega-project: Building A Redundant Asterisk@Home System.

Overview. Now that we have Asterisk@Home 2 up and running, we need to do a couple of additional Linux things. First, we need to add the Linux bluetooth software to your system. And then we need to install WebMin, the Linux answer to the Swiss Army knife. Once we get our software configured, we’ll plug in our dLink DBT-120 bluetooth adapter and take our new system for a spin. We’ll test our ability to detect other bluetooth devices such as a bluetooth-enabled cellphone or a bluetooth headset. Then we’ll be ready for Chapter 3 where we’ll actually start telling our primary Asterisk server when we’re IN and when we’re OUT.

Installing Linux Bluetooth Software. For the remainder of this article, we’re going to be working exclusively with your new Asterisk@Home 2 beta system so, when the instructions say to do something on your Asterisk box, that means on your Asterisk@Home 2 beta system not your primary Asterisk@Home PBX that is handling your phone calls. Log in to your new Asterisk box as root.

For those using Asterisk@Home 2.0 beta 6 or later, the bluetooth software comes preinstalled with CentOS 4.2 so you can skip this next download and install step. If you have an older version of the Asterisk@Home 2.0 betas, then, to download and install the bluetooth software for Linux, issue the following command:

yum install bluez-libs bluez-pin bluez-utils bluez-hcidump bluez-utils-cups

Your system will alert you that there are some dependencies and ask if you want to download them as well. Answer ‘yes’ and the download and install should proceed without incident. Once everything finishes, plug in your dLink DBT-120 USB bluetooth adapter and restart your system: shutdown -r now. When your system reboots or if you’re running beta 6 or later, log in as root and issue the following command:

/etc/init.d/bluetooth restart

You’ll probably be told that your system couldn’t stop bluetooth (because it wasn’t running) and then it’ll restart. Now let’s make sure everything is running that should be:

/etc/init.d/bluetooth status

You should see messages that look like this:

hcid (pid somenumber) is running ...
sdpd (pid somennumber) is running ...

If you’re alerted that some other application isn’t running, we don’t care. Now let’s be sure the system has found your bluetooth adapter:

hcitool dev

You’ll get a response telling you the system found device hci0 with the MAC address of the adapter. If you have a bluetooth-enabled cellphone, go get it now and be sure it’s on with bluetooth-enabled and not in sleep mode. Now issue the following command:

hcitool scan

The system will whir away for a few seconds and then will report back the bluetooth name assigned on your cell phone. It’s probably your name. Didn’t know you were advertising, did you? Note: you can’t do this with a bluetooth headset, but we’ll get to that in a minute. There are two ways to automatically start the bluetooth software when you boot CentOS. We’ll show you both, but you’ll probably want to install WebMin for future use anyway. The quick and dirty method is to run setup while logged in as root. Use the down arrow key to highlight System Services and press ENTER. Now use the down arrow to move down to bluetooth. Then press the space bar. Tab to the OK button and press ENTER. Now tab twice to the Quit button and press ENTER. Restart your server and you’re all set: shutdown -r now.

Let’s take a bluetooth break and install WebMin. We’re doing this primarily to show a simpler way of auto-enabling bluetooth at startup, but you’ll find lots of other uses for WebMin down the road.

Installing WebMin. There are lots of ways to install WebMin. We prefer the easy way which is to issue the following commands at a Linux prompt after logging in as root. Note: WebMin updates come out all the time. If you want to be sure you start with the latest and greatest version, go to their web site first and write down the number of the current version. Then substitute it below when issuing these commands:

cd /root
mkdir webmin
cd webmin
wget http://unc.dl.sourceforge.net/sourceforge/webadmin/webmin-1.240-1.noarch.rpm
rpm -Uvh webmin*


WebMin runs its own web server on port 10000. To start WebMin, issue this command: /etc/webmin/start. You access it with a web browser pointed to the IP address of your Asterisk box at that port address, e.g. http://192.168.0.108:10000. The login name is root. Then type in your root password and press enter. The main WebMin screen will display. Before we forget, we need to also make one change to the new Asterisk@Home configuration to avoid problems down the road. The default RTP listening ports for Asterisk@Home are 10000 to 20000 so there’s a conflict on port 10000. We just need to change it to 10001. Log in as root and, using an editor, call up the rtp.conf file: nano /etc/asterisk/rtp.conf. Now change the rtpstart port from 10000 to 10001 and save the change: Ctrl-W, Y, and press Enter.

Now back to WebMin. From the Main Screen of WebMin, click the System button and then the Bootup and Shutdown link. Find bluetooth in the list of applications and click on it. The Action Details screen should show that bluetooth is running. Now click the Yes button beside "Start at Boot Time?" and then click the Save button to reconfigure your server. That wasn’t hard, was it? We can stop WebMin now by issuing the following command: /etc/webmin/stop. No need to waste processing cycles for a tool we’re not using.

Configuring a Bluetooth Headset for Proximity Detection. As we mentioned in our first article, our device preference for the proximity detection project is a bluetooth-enabled headset. The two least expensive ones (about $30 including shipping) are the Plantronics M3000 if there are any left and the IOgear GBE201W7 from Buy.com. Remember, we’re going to be using the headset primarily as a "key" to tell Asterisk when we’re home and when we’re not. Cadillac charges about $400 for their electronic keys so this one’s a bargain. You obviously can use it for other stuff, too, but for now let’s get our "key" working. The hardest part of this drill is figuring out the MAC address of the headset. If yours happens to have 12 hex digits written on the box, lucky you! Otherwise, the only trick here is to put your headset in discovery mode. On the M3000, you activate discovery mode by simultaneously holding down the UP volume and Command buttons for about two seconds. When the light begins alternating between red and green, let go. With the IOgear unit, turn off the device and then hold down the multi-function button for about nine seconds. The red LED will turn on and then the blue LED will begin rapidly flashing. If only the blue LED is lit, you screwed up. Try again. Now from a Linux command prompt in close proximity to your headset, issue this command:

hcitool scan

You’ll get a response that looks something like one of the following:

Scanning ...
00:03:89:43:84:E2 M3000 by Plantronics
00:0e:a1:21:39:fc IOGEAR BT Headset


Write down the MAC address of your headset including the colons between the hex code digits. This is the notation we’ll be using to actually check whether your headset is within range. Here’s the command so try it out substituting the MAC address of your headset for mine below:

hcitool name 00:03:89:43:84:e2

Now run the command again and change the last digit of the MAC address to another hex number:

hcitool name 00:03:89:43:84:e6

There are two important points here. First, with the hcitool name syntax, the headset does not have to be in discovery mode which is just what we need for our proximity checking. And second, you’ll notice you got the device name when hcitool could find the headset and a null string when it couldn’t. So we can use a simple Linux bash script to determine whether your headset is within range by just checking whether a null string is returned to the above query. Wouldn’t you think some commercial developer could have figured this out and included it in their highly touted Proximity Checking products? Apparently not.

We’ll leave you with a simple script today to play with until we tackle Chapter 3. This script implements what we’ve just demonstrated. All you need to do is create a new file in the /tmp directory of your new Asterisk server: nano whereib. Then cut-and-paste the code below and modify the MAC address shown for deviceid= to match your headset. Then save the file: Ctrl-X, Y, and press enter. Now make the file executable: chmod 755 whereib. And then run it: ./whereib. Once it’s running, turn on your bluetooth headset and walk around your home or office. Then come back and check the display. It should report when you’re in range and when you’re not. I think you’ll be pleasantly surprised by the range of your bluetooth devices so the basement furnace room may not be a bad location for your new server after all. You can end the script by pressing CTRL-C. In our next article, we’ll wrap up the bluetooth proximity series with Chapter 3 where we’ll get everything connected up to your primary Asterisk server which will handle the actual call forwarding when you’re away from your home or office.

#!/bin/bash
# Syntax: ./whereib
# On a clunker PC it takes about 20 seconds for a test to fail
# and it takes about 5 seconds for a test to succeed, i.e. BT MAC found

count=0
while [ $count -lt 1 ] ; do
hcitool name $deviceid > /tmp/$devicename
if [ -s /tmp/$devicename ] ; then
echo $devicename IN RANGE ;
date
else
echo $devicename OUT OF RANGE ;
date
fi
sleep 7
done


Some Interesting Bluetooth Reads. Here are a few articles to give you an idea where all the bluetooth proximity detection stuff is headed. If it’s good enough for Bill Gates’ home, it’s good enough for yours.

Other Asterisk Tutorials. There are numerous additional articles in this Asterisk HOW-TO series to keep you busy. You can read all of them by clicking here and scrolling down the page. We recommend reading at least the first four or five articles from the bottom up so that the learning curve is less painful. Then you can skip around to your heart’s content.

This is the first in a series of articles that will provide step-by-step instructions for implementing Bluetooth Proximity Detection. We’re going to focus on using it with Asterisk®@Home, a terrific PBX which also happens to be free. But your imagination is really the only limitation. At the very least, when we’re finished, you’ll be able to walk out of your home or office carrying your bluetooth phone or headset and have your Asterisk server automatically transfer your incoming calls to your cellphone. And, when you return carrying your bluetooth phone or headset, Asterisk will automatically cancel the call transfers and reactivate delivery of incoming calls to the designated phones in your home or office. As simple as this concept may sound, the devil is in the details. So we want to spend today warning you of all the minefields that lie ahead and telling you what hardware you’ll need to make things work. If you hurry, you can implement the whole system for just over $50, and we’ll show you how to do it without even owning a bluetooth cellphone. In subsequent articles, we’ll put the pieces together and get a basic system working. Then we’ll add more bells and whistles and give you some implementation and deployment suggestions. You’ll quickly come to appreciate how Bluetooth Proximity Detection can be used to implement all sorts of other features. When we’re finished, you’ll also appreciate the potential of bluetooth to revolutionize the workplace. And it goes far beyond your phone system. Imagine an automated IN/OUT message board in businesses such as real estate or the advertising potential to tailor TV display ads in stores based upon not only your presence but also the type of cellphone you are carrying. Your office can even kiss its old punch clock goodbye when we’re finished. For those with new Cadillac or Mercedes automobiles, you can unlock your car and start it just by approaching the vehicle with your "key" still in your pocket. So where do we start?

NOTE: This article has been updated to take advantage of TrixBox, freePBX, and the iPhone. For the current article, click here.

Overview. The basic idea behind proximity detection is that we run a software application on a computer to "watch" for approaching people. We then want it to do something when you (or a customer) gets within range. How do it know? Well, in our case, this is where Bluetooth comes in. Unlike motion detectors which can’t tell the difference between a human and a gorilla, bluetooth devices all have a unique MAC address just like a network card. And most bluetooth devices also have a name. So long as the bluetooth device is configured to advertise its presence, we can detect when it is within range and when it’s not. That’s the second major difference between bluetooth and traditional motion detectors. Ever been in a public restroom or an office when all the lights went out because everybody was sitting too still for too long? So motion detectors have some limitations. Bluetooth doesn’t. In case you’ve been living under a rock for the past six years, bluetooth is a wireless communications protocol that uses short range radio frequency to connect devices into wireless personal area networks (PANs). The most common Class II 2.5mW devices have a range of 32 feet (10 meters). Class I devices have a range of up to 100 meters. Most bluetooth cellphones, headsets, and computer peripherals such as mice and keyboards are Class II devices. If you want more background, go here.

Prerequisites. For our proximity detection project, we’re going to connect a bluetooth network adapter to an Asterisk@Home box and make it our master. That simply means we’re going to use this network adapter to look for other bluetooth devices within range. The only limitation is you can’t have your Asterisk@Home box shoved in a closet in the basement if you want this to work. It will need to be within 30 feet or so of where you’ll be when you’re at home or in the office. If this doesn’t work for you, then here’s an alternative. Just get in the habit of putting your cellphone or bluetooth headset down near your Asterisk@Home box when you’re "in" and take it with you when you’re "out." Many offices, particularly in the real estate business, have a receptionist with agent mailboxes immediately beside or behind the receptionist desk. Just put your Asterisk box with its bluetooth adapter under the receptionist’s desk and leave your cellphone or wireless headset in your mailbox whenever you return to the office. The adapter we recommend which is quirk-free is dLink’s DBT-120. You can find them on the net for about $30, but you can usually beat that price by watching the Sunday circulars for computer and office depot/max stores in your area … if you don’t mind mail-in rebates. But, do you really want the PBX for your whole office sitting under the receptionist’s desk? Probably not. But don’t worry, we’ve got some other tricks up our sleeve so keep reading.

We keep mentioning a headset so we won’t keep you in suspense any longer. You don’t need a bluetooth cellphone to make our proximity detection project work. A bluetooth wireless headset works just as well. In fact, it works better! And you’ll have a great addition to your computer system and cellphone as an added bonus. Cellphones have a nasty habit of putting themselves in sleep mode very quickly when not in use to conserve battery power. The only problem is that most, if not all, cellphone makers turn off the bluetooth adapter when they activate sleep mode because they’re all so short-sighted that the only thing they think you use bluetooth for is to talk to your wireless headset or exchange files with your PC. Stupid! Bluetooth headsets on the other hand are always on listening for a call. The one we like has a rated standby time of 200 hours between battery charges so it’s perfect for this project. These devices typically cost anywhere from $50 to $100 but, if you hurry, there’s a vendor selling our favorite, the Plantronics M3000, for under $20. Here’s the link at PriceGrabber. Don’t wait. They’re never this cheap, and this vendor only has 50 of them. And Buy.com has a similar unit from IOgear for about the same price once you factor in the cost of shipping. Will you need to wear your bluetooth headset and look like a Nerd to make this work? Not at all. Just turn it on, stuff it in your pocket, and call it a key.

Now let’s address the computer issues. First, your machine obviously needs USB adapter support so you have a place to plug in your bluetooth adapter. Second, we need a machine that can run software that can detect bluetooth devices. Having spent a week scouring the Internet and testing various products which touted their bluetooth proximity detection, let me save you some time. If you are fortunate enough to have a Sony Ericsson phone with bluetooth, some of the commercial products such as BluePhoneElite for the Mac or Salling Clicker for Mac or Windows work great for proximity detection. There’s even an open source product, Romeo for the Mac, that works. If you have a single-tasking Palm device including the Treo 650 cellphone, don’t waste your time. And bluetooth headsets aren’t detected at all by any of the products. This is primarily because proximity detection was considered a gee-whiz extra in most of these products so it’s not implemented very well. The good news is that, if you happen to have a bluetooth cellphone that does work with one of these products, it might make proximity detection more practical because you could handle the proximity interaction with your desktop machine instead of with your telephone system’s PBX. But, who cares. We just want it to work.

So where does that leave us on the computer front? The bottom line is you’re going to need a Linux machine and a fairly current version of the Linux operating system to get the bluetooth tools installed that we need. As luck would have it, the new Asterisk@Home 2.0 beta release works great … and it’s free. And it automatically installs CentOS/4, the free knock-off of RedHat’s commercial Enterprise Linux 4. Because Asterisk@Home is free and will run on any old clunker PC, you may want to install the Asterisk@Home 2.0 beta on a dedicated machine and just use it for proximity detection. This solves the colocation problem with your main PBX, and it has the added benefit of reducing the load on your primary Asterisk server. The other terrific benefit of this approach is you’ll have a hot standby system for your main PBX, and we’ll integrate that into our tutorials one of these days, too. When your one and only Asterisk@Home box dies, do you really want to be without phone service? Keep in mind that proximity detection also takes some horsepower because we’ll be running a script once a minute to see who’s in and who’s not. And, no, Asterisk@Home 1.5 won’t work. Believe me, we’ve tried and it was just about as frustrating as trying to use a Treo 650 for proximity detection. A total bust!

Well, that covers the basics and provides you the information you’ll need to start assembling the pieces for the proximity detection project. We’ll leave it to you to get your bluetooth hardware ordered and to get your Asterisk@Home 2.0 beta up and running before moving on to Chapters 2 and 3.

Some Recent Nerd Vittles Articles of Interest…

This is the second in our two-part roundup of the best unlimited calling plans for Asterisk®. You can read the first installment here. The number of options for Asterisk residential users wanting an unlimited international calling plan has fluctuated between zero and one depending upon how brave you were in dealing with BroadVoice's Terms of Service. Frankly, we've pretty much written off BroadVoice's so-called unlimited international calling plans for residential use because of the number of backbilling complaints logged on the Voxilla Forum.

Finally, there's not only some competition but also a ray of hope. Axvoice Inc. has recently announced two new unlimited residential calling plans with full bring-your-own-device support for Asterisk. An $18.99 plan provides unlimited calling within the U.S. and Canada. Spending $4 more buys you unlimited calling within the U.S., Canada, and all or parts of the following countries: Buenos Aires, Argentina; Australia; Chile; Denmark; France; Germany; Hong Kong; Ireland; Israel; Italy; Monterey and Mexico City, Mexico; Netherlands; Norway; Moscow and St. Petersburg, Russia; Singapore; South Korea; Spain; Sweden; Taiwan; United Kingdom; and Vatican City. As usual, mobile, premium, and special numbers are not included in the program. The fine print places a 4,000 minute usage cap on residential service before it is considered business use. This seems more than reasonable considering that such a number gives the residential user over two hours of free calls per day, every day of the year. Stated another way, the $22.99 international calling plan provides up to 4,000 minutes of calling per month to 24 countries for an average per minute cost that works out to just over a half cent a minute. Compared to the BroadVoice approach which leaves you guessing (at your financial peril) what the usage caps are for their various unlimited* calling plans, the Axvoice approach is a breath of fresh air. If you need to stretch your minutes and receive significant numbers of incoming calls, you can supplement your all-you-can-eat plan with a BYOD plan featuring unlimited, free incoming calls for $8.99 a month. Or you could simply add another all-you-can-eat plan for an additional 4,000 minutes. You can't beat the price, and all of Axvoice's plans include two free incoming DID numbers from a broad selection of area codes within the U.S. and Canada. Currently Axvoice is waiving the $9.99 setup fee for bring-your-own-device users. How do the calls sound? Ours sound great. Your mileage may vary. But you can try it for yourself with their 30-day money-back guarantee. So you really have nothing to lose by trying out their service. And, if you stay with Axvoice, all of their bring-your-own-Asterisk plans are month to month so there are no long term commitment issues.

The equally important question for Asterisk users is always the same. Does it work well with Asterisk? Again, the good news is that, once configured, Axvoice and Asterisk are a match made in heaven. In fact, Axvoice uses Asterisk at their end so you wouldn't expect anything less. Axvoice is a relatively new company, but you'll notice some marked differences from BroadVoice if you've endured BroadVoice support with their long waiting times and frequent disconnects while on hold. I think I was one of the first home users that wanted to set up Axvoice service with Asterisk so I can't say it was painless, but I can pretty much assure you that the process will be close to painless for you. We wrote down the answers! At least as this article goes to press, when you sign up on their site, you won't be greeted by much of any documentation pertaining to Asterisk configuration. All that's about to change if you keep reading here. Again, they're a new company, and we pretty much knew going in that it probably would be a bumpy ride. At least for the first three days, we weren't disappointed. The good news is that email inquiries always were returned with a helpful answer within an hour or so. And calls to their toll-free number were answered in less than a minute both during the business day and at night. You really couldn't ask for much more.

I think it would be fair to say that Axvoice's typical customer up to now has pretty much been like Vonage's, dumb as a brick about technology and looking for a good deal on an unlimited VoIP calling plan. So, when you call for customer service, the typical first response is to inquire about your router/firewall. And they fully expect you to answer by saying, "What's that?" Unlike Vonage, Axvoice fully supports Asterisk with an open SIP configuration. But it took some perseverance to get to the person who knew the answers. Some of the first-level customer service reps were having a difficult time both with English and trying to understand that we, too, had an Asterisk server ... at our home. "Really?" was a typical response. Once we got by all of that, the Asterisk configuration information was detailed and to the point. The other wrinkle we experienced had to do with their support for the G.729 codec. We only have one license which may have been the problem. Outgoing calls with G.729 worked fine, but incoming calls that first hit our AutoAttendant and then get transferred to a default extension lost the outgoing part of the audio. This may be because Asterisk treats the two voicepaths (incoming and the answering extension) as two separate channels requiring two, rather than one, G.729 license. When payday rolls around, we'll splurge for another $10 G.729 license from Digium®, and let you know the results. In the meantime, the default ulaw codec works great.

Configuring Asterisk Trunk for Axvoice. Assuming you decide to take the dare and sign up for a test run, what you're going to need to get Asterisk running with Axvoice is your account ID (not a phone number) and your password. If you've followed along in our previous Asterisk@Home tutorials, then the process to set up Axvoice will be familiar. Be sure UDP ports 5004-5082 and 10000-20000 are opened on your firewall and pointing to your Asterisk server. We'll create a trunk for Axvoice and then add Axvoice to our dialplan to take advantage of their calling plan. Using the Asterisk Management Portal, goto AMP->Setup->Trunks and Add a New SIP Trunk. You can leave the CallerID field blank since this is configured on Axvoice's web portal. For maximum channels, enter 2. For Outgoing Dial Rules, enter the following but be aware that this dialplan will allow international calling to many country codes that are NOT free under the Axvoice unlimited international calling plan. If this is a problem in your household, you'll need to make your dialplan much more country-code specific.

1+NXXNXXXXXX
1NXXNXXXXXX
011X.

For Outgoing Settings, name your trunk axvoice and then enter the following Peer Details substituting your username (3 entries) and password (1 entry) where appropriate:

allow=ulaw
authname=yourusername
canreinvite=no
context=from-sip-external
defaultip=sip.axvoice.com
disallow=all
dtmfmode=rfc2833
fromdomain=sip.axvoice.com
fromuser=yourusername
host=sip.axvoice.com
insecure=very
nat=yes
secret=yourpassword
type=friend
user=phone
username=yourusername

Leave the Incoming Context and Details blank. Enter your registration string as follows using your actual username and password. Then save the trunk and click the red bar to reload Asterisk.

yourusername:yourpassword@sip.axvoice.com

You will recall from our previous article on the subject that the [from-sip-external] context passes incoming calls to our Stealth AutoAttendant for processing. So Axvoice incoming calls will be handled in the same way as incoming calls from your other PSTN and VoIP trunks.

Adjusting Outgoing Dialplans for Axvoice. To adjust your dialplans to take maximum advantage of Axvoice, choose AMP->Setup->Outbound Routing. Then for your Local, InState, TollFree, and US dialplans, edit each of them and Add the SIP/axvoice trunk. Then use the arrow keys to move Axvoice to the top of each list. You'll also want to add a new International route with the following Dial Pattern: 011X. which designates all calls beginning with 011 and at least two more digits as international calls. Now choose SIP/axvoice as your one and only trunk for international calls. If you want a fallback trunk, Voxee would be our recommendation. They have incredibly low per minute rates. Now save your trunk settings and leave the international dialplan at the bottom of your list of Dialplan Routes.

Making a Test Call Using Axvoice. To be sure everything is working swimmingly, start up Asterisk in interactive mode using the Command Line Interface (CLI) so that you can actually watch what's happening when calls are placed and received. This works best if you connect to your Asterisk server through SSH from a Mac or PC. SSH comes with every Mac and the syntax is simple: ssh root@AsteriskIPaddress. If you're still chained to Microsoft, download Putty from the Mother Country, and you can do the same thing using a Windows machine. Once you're logged in as root, issue the following command: asterisk -r. Quit ends your Asterisk CLI session, and exit logs you out of your SSH session. Now issue the command: set verbose 10 to get maximum information. Then place a U.S. or international call and watch what happens. You should see something similar to the following which shows that the call was placed using the new axvoice trunk:

-- Called axvoice/18435551212
-- SIP/axvoice-2cbf is ringing


Asterisk: The Future of Telephony. O'Reilly Publishing finally released a great book on Asterisk last week. The book is available directly from the publisher or from Amazon. In addition, O'Reilly is providing the PDF version at no cost under a Creative Commons License. It's a carefully written book and an easy read for beginners as well as those with lots of Asterisk miles under the belt. It does not cover Asterisk@Home. And what more can we say about O'Reilly Publishing. O'Reilly is simply one terrific company that just keeps getting better! Some other Asterisk books are now available as well. You can read all about them here. And at least one Asterisk@Home book is currently under development ... not from us. We don't do books. You can always tell when a new technology is really taking off by watching for book publications. For those that have been following these tutorials, I think you understand why Asterisk is finally earning its rightful place in the limelight.

Other Tutorials. There are numerous additional articles in this Asterisk HOW-TO series to keep you busy. You can read all of them by clicking here and scrolling down the page. We recommend reading at least the first four or five articles from the bottom up so that the learning curve is less painful. Disclaimer: As we have mentioned previously, we typically sign up for referral credits when we try out a new VoIP service. If our referral credit links give you heartburn, don't use them. We do it to help defer costs of this column. It doesn't influence our views of the providers as you probably can tell from our discussions of BroadVoice. They too have a referral credit program, and we still don't like their current terms of service. Fortunately, thanks to Axvoice, international callers finally have another viable unlimited calling plan option.

Coming Thursday: AsteriDex -- The Poor Man's Rolodex. We're excited about our latest free software offering and think you will be, too. AsteriDex is a web-based application that lets you store phone numbers of all your friends and business associates in a simple-to-use MySQL database. But, it's much more with Asterisk! Just call up the application in your favorite web browser from anywhere and click on the contact you want to call. AsteriDex will first call you at the number you've designated for this contact, and then AsteriDex connects you to your contact through an outbound call made using your Asterisk server. For those lucky enough to have GrandStream's GXP-2000 IP phone with AutoAnswer, you can even configure AsteriDex to automatically activate the speakerphone and then place the call to the contact you've selected. In short, it works exactly like Microsoft's TAPI software without the configuration nightmare or your favorite (required) Micro$oft bloatware. Installation and configuration for your Asterisk@Home system is a snap and takes less than 10 minutes. It also works with vanilla Asterisk running the Asterisk Management Panel (AMP) software. Someone is probably saying, "Can't you do the same thing with CRM?" Well, yes and no. If you need full-blown contact management, then sure you can. Just be sure to save an extension for the account you are using to log in using the format sip/204. CRM will dial out using your default Asterisk outbound dialing rules, and it works great. But be prepared to spend several months getting all your contact information entered. AsteriDex is quick to implement and simple to use because it only does one thing: places calls for you using a web interface to all your favorite callees. The software is distributed without charge pursuant to the Creative Commons License.

The real payoff for installing that fancy Asterisk PBX in your basement comes when you’re on the road and want to make free phone calls either to or through your home system. There are probably a million ways to do this. Most of them are painful. This is particularly true with SIP-based telephones and all the problems associated with configuring NAT and firewalls and STUN servers. So, as usual, we’re going to take the low road and do things the easy way. We’ll give you one solution that really works … every time! When we’re finished, you’ll have a device about the size and weight of a pack of cigarettes to carry on your trip. And all you’ll need when you arrive at your destination is a 10/100 network connection with a cable and a plain old telephone with a plain old phone cord. Once you connect the two devices, you simply pick up the phone and dial calls just as if you were sitting in your kitchen at home. And incoming calls work just as if you had added another phone in your upstairs bedroom. It’ll even flash at you when you have voicemail waiting. For those with kids in college, this is the perfect addition for the dorm room if you want to avoid cellphone hell. And the one-time, non-recurring cost: under $100.

To make all this magic work, you’ll need to purchase Digium®’s S101I, affectionately known as the IAXy Version 2, a NAT-transparent, FXS device providing a POTS telephone interface to your Asterisk® PBX using an IAX connection. You can buy one directly from Digium, the makers of Asterisk, by going here. Be sure you order it with a power adapter for your particular country. Buying the device directly from Digium is an especially good idea because you can call them should you ever have a configuration problem. Hopefully you won’t after you finish reading this article.

IAXy IP Configuration. Once your IAXy arrives, it’s configuration time. First things first. You can download Digium’s installation guide or just keep reading for a bit more hand-holding. The device needs to be connected to a network that has a DHCP server so that an IP address is automatically handed out to the IAXy when you plug it in. Before you power up the unit, plug in a plain old telephone and connect the IAXy to your router using a 10/100 network cable. Then plug in the power adapter. The hardest part of this drill is probably figuring out what IP address was assigned to the device. On most routers, you can use a web browser to access the router configuration. Usually there’s an option to display Attached Devices. Typically, the last device you plug in gets the highest IP address so take a look and write it down. Write down the MAC address of the unit, too. Luckily, you can ping the IAXy to see whether it’s alive. So, from your desktop machine, ping the IP address you wrote down, e.g. ping 192.168.0.123. If you don’t get a reply, that’s the wrong number. Try another one. If you do get a reply, unplug the IAXy and ping the same IP address again. If you get a reply, that’s obviously not your IAXy because it’s unplugged so start over. If you don’t get a reply, you’ve got the right number. Now plug it back in and do it one last time to be sure you still have the same number. Before you forget, go into your router configuration now and permanently assign this IP address to the IAXy. It’s usually named something like Reserved IP Table, and you’ll find it in the LAN IP setup screen on most routers. Finally, if your Asterisk server is behind your router/firewall, you’ll need to open UDP Port 4569 on your router and map it to the private IP address of your Asterisk server, not your IAXy. On most routers, you’ll find port mapping under a heading of Services or Rules. IAX is not a common protocol so you may need to create it. Just name it IAX, specify UDP as the protocol, choose port 4569, and save your settings. Then add a rule that maps this IAX port to the private IP address of your Asterisk machine. Whew!

IAXy Provisioning Utility. One more knuckle drill, and then it’s pretty smooth sailing. This isn’t the Windows world so you’ll need to compile the IAXy provisioning utility on your particular flavor of Linux. We use this utility to actually configure the IAXy device. Nothing here is difficult. Just follow the steps in order. Go to your Asterisk server, switch to root user access, and issue the following commands to download and compile the source code for the IAXy provisioning utility:

$ cd /usr/src
$ export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
$ cvs login (the password is anoncvs)
$ cvs co iaxyprov
$ cd iaxyprov
$ make
$ chmod 775 iaxyprov


Configuring Asterisk to Support IAXy. We need to take a break here an actually set up an extension for your IAXy device to use. If you’re using Asterisk@Home or at least the Asterisk Management Panel, go to AMP->Setup->Extensions and click Add Extension. Enter the following using whatever extension and password you prefer:

phone protocol: IAX2
extension number: 222
extension password: 1234
full name: IAXY

If you want to activate voice mail for this extension, choose Voicemail and Directory Enabled and make the VoiceMail Password match your extension password. If you want email delivery of your voicemail messages, fill in the appropriate blanks. Then save your settings by clicking Add Extension. Click the Red Bar to reload your Asterisk configuration.

IAXy Configuration File. The next step is to decide where you will be using your IAXy device. If it will always be outside your firewall, then we only need to configure the device to use the public IP address of your Asterisk machine. If it will always be inside your firewall, then we just need to assign the private IP address of your Asterisk machine (assuming it’s behind your router/firewall). If you want to be able to move the device inside and outside your firewall, then we need to assign two IP addresses: the public and private IP addresses of your Asterisk server. The only other gotcha here is that if your Internet Service Provider assigns you a dynamic IP address, you will need to reconfigure the public IP address of your IAXy device every time your ISP changes your dynamic address. There is no DynDNS support for the IAXy at this time.

We’re going to initially configure your IAXy device for use inside your firewall where your Asterisk server also lives. Then we can play with it to make sure it’s working. Then we’ll walk through the steps to change the configuration. The IAXy configuration file and the provisioning utility we built earlier live in the /etc/src/iaxyprov directory on your Asterisk machine. It’s a good idea to switch to root user access whenever you work here just to keep all the files protected from snoopy people. Let’s make a duplicate of the IAXy config file just in case something gets screwed up: cp /usr/src/iaxyprov/iaxy.conf.sample /usr/src/iaxyprov/iaxy.conf. Now open iaxy.conf using your favorite editor: nano iaxy.conf. It should look like this:

;
; IAXY Provisioning description
;
dhcp
;ip: 216.207.244.130
;netmask: 255.255.255.192
;gateway: 216.207.244.129
codec: ulaw
;codec: adpcm
server: 192.168.0.1
;altserver: 192.168.0.2
user: myuser
pass: mypass
register
;heartbeat
;debug
;
; Feature tuning (default is all enabled)
;

You only need to change three items in the file. For server, replace this IP address with the private IP address of your Asterisk server. For user, plug in the extension number you assigned to the IAXy. And for pass, enter the password you assigned to this extension. Save your changes and exit from the editor: Ctrl-X, y, enter. To actually load the configuraton into the IAXy, issue the following command using the IP address assigned to your IAXy:

./iaxyprov 192.168.0.100 iaxy.conf

If there is no error in your config file, you will get a screenful of feedback from the IAXy device within about 10 seconds. Just disconnect power to the IAXy and then reconnect it, and you’re in business. If you get nothing on the screen after issuing the above command, then there is an error in your config file or your IAXy device is not connected to the network with the IP address shown in the iaxyprov command line you entered. Check your work and try again.

Reprovisioning the IAXy. Loading a new configuration into the IAXy is easy now that you know the drill. It’s always a good idea to reset the IAXy to its factory default settings before loading a new configuration. Here’s how:

Once the IAXy is restored to its factory defaults, you simply repeat the configuration steps above to reprovision the unit. To use the unit on the outside of your firewall, edit the iaxy.conf file and change the server IP address to the public IP address of your Asterisk server. Save your change and then load the new settings with the same iaxyprov command line we used above (assuming your IAXy is still plugged in behind your firewall). Note that you will not be able to test the device behind your firewall so take the unit to a friend’s house (with broadband) and try it out with any plain old telephone. If you want to use the IAXy both inside and outside your firewall, we’ve had good luck simply plugging in both IP addresses. Leave your private IP address in the server line, and uncomment the altserver line by removing the leading semicolon. Then enter your public IP address here, and reprovision the unit as previously explained.

Asterisk Dialplan Quirk. If you’re used to ringing all your phones for incoming calls, you’ll quickly discover that the addition of an IAX phone device complicated things a bit. You can’t mix and match device types in a Dial command. For example, exten => 1234,1,Dial(IAX2/277&SIP/204,20,r) won’t work. There’s a simple way to get around this limitation. Use a ring group which includes both SIP devices and IAX2 devices such as the IAXy. The syntax for the Dial command would look like this assuming 299 was your Ring Group: Dial(local/299@from-internal,20,m).

Activating MD5 password encryption. One great addition to the version 2 IAXy was support for MD5 password encryption. If you’ll be using the device in public places or hotels, you’ll want to use this. One quick change in your Asterisk configuration is all that’s required. Using AMP->Maintenance->Config Edit, edit the iax.conf file. Just add the following line in the [general] section of the file:

auth=md5

Save your change and restart Asterisk, and you’re all set to hit the road. Enjoy!

Coming Attractions. We’ve been working on another web application using Asterisk that builds on our previous CallMe application. For lack of a better name, we call this one The Poor Man’s Rolodex. When it’s soup, it will let you create a protected web page with the names and phone numbers of all your favorite people and places. Making a selection on the web page passes a command to a PHP script that talks to your Asterisk PBX. It first rings one or more phones in your home or business. Only when the call is answered does it place an outgoing call to the person or place you selected on the web page. It then automatically connects them to the phone you picked up. If you happen to have GrandStream’s GXP-2000, you can even configure that phone to go off-hook and turn on the speakerphone when the incoming call from your ‘Rolodex’ arrives. In short, this little ditty will give you everything Microsoft’s TAPI gateway provides without any of Microsoft’s proprietary baggage.

Some Recent Nerd Vittles Articles of Interest…

One of the really terrific features of Asterisk® is it’s ability using DISA (Direct Inward System Access) to provide dial tone to an incoming caller. This allows a caller to Phone Home and place outgoing calls through a remote Asterisk server to take advantage of all those VoIP cost savings we’ve been discussing ad nauseum. You obviously need to be thinking about security before you implement DISA but, properly secured, DISA is one of the most powerful functions of your Asterisk PBX so why not use it to your advantage. But there are some wrinkles. Suppose you’re traveling in a foreign country that charges a $14 minimum for any completed call to the U.S. regardless of duration. Or you may just be at a neighbor’s house and want to make a quick call on your nickel to check on Aunt Betty in Paris. Or you may be on a Nextel free incoming call plan and don’t want to burn up your cellphone minutes placing outgoing calls directly with your cellphone.

Several weeks ago, we provided a quick and dirty HOW-TO on activating a DISA callback using a web browser and entering a specific command to your Asterisk@Home’s web server. But there may be folks that don’t want the security risks associated with supporting a web server. So today we want to revisit our original Phone Home column and give you three different ways to implement DISA. The three methods are the following: (1) the AutoAttendant, (2) the CallMe Web Interface, and (3) the One Ringy-Dingy. These obviously can be mixed and matched to meet your own specific requirements.

AutoAttendant DISA. The simplest DISA implementation is to add an option to your AutoAttendant. With this option, you phone home, pay the costs of the call, and while still connected make another call through your Asterisk server by picking the DISA selection when your AutoAttendant plays. You’ll be prompted for a password and, after entering it correctly, Asterisk will provide dialtone for your use. The drawback of this option is obviously the cost, if any, of the call to your home base. If that’s not a problem, then this is a great solution. And it’s very easy to implement. Take a look at the [from-external-custom] code in our Securing Your Asterisk@Home PBX article for all the details. But basically you only need to add a couple of lines to your AutoAttendant to support DISA. Choose the number that people will press to activate DISA and pick a very secure password, and you’re all set. Assuming the number to press is 4 and your chosen password was 1234588, here’s how to set up the AutoAttendant code to implement DISA. It doesn’t get much easier than this.

exten => 4,1,Authenticate(1234588)
exten => 4,2,Background(pls-wait-connect-call)
exten => 4,3,DISA(no-password|from-internal)


CallMe Web Interface. We’ve put together a little web application (actually a PHP script) so that, using a web browser on the road, you can tell your Asterisk server to call you and provide dialtone to any number you specify. The only prerequisite here is that we don’t want to sell the farm, i.e. provide free dial tone service and unlimited international calling for all the world’s hackers and crackers. We also don’t want to have to go through a bunch of authentication steps to access the web site and put the call in motion. So here’s the design. We have a PHP script which you can download here. It needs to be renamed to callme.php. Then copy it into the /var/www/html directory on your Asterisk server. You’ll also need to tell your firewall/router to route HTTP or port 80 traffic to the internal IP address of your Asterisk server. This is usually done under the Services or Rules menus on most routers. You’ll want to specify that all port 80 traffic be allowed through the firewall all of the time. Be sure you’ve changed ALL of your Asterisk passwords before you do this!

To use this script from the Internet, you’ll probably want to have to have a more permanent fully-qualified domain name associated with your Asterisk server. We explained here how to do this using dyndns.org. If you use a SIP provider with your Asterisk server, the syntax is as follows: http://asterisk.dyndns.org/callme.php?number=sip/bv/4045551212 where asterisk.dyndns.org is the fully-qualified domain name for your Asterisk server and 4045551212 is the area code and number where you wish to accept a call with dialtone, and bv is the outgoing trunk name of your SIP provider. If you use an IAX provider with your Asterisk server, the syntax is as follows: http://asterisk.dyndns.org/callme.php?number=iax2/goiax/14045551212 where asterisk.dyndns.org is the fully-qualified domain name for your Asterisk server and 14045551212 is the all-important 1 followed by the area code and number where you wish to accept a call with dialtone, and goiax is the outgoing trunk name of your free IAX provider. Nothing else needs to be changed. To dial a local extension, use this syntax: http://asterisk.dyndns.org/callme.php?number=sip/204 where asterisk.dyndns.org is the fully-qualified domain name for your Asterisk server and 204 is the local number to ring. Beginning on the first ring, Asterisk will start prompting for a password. It doesn’t care whether the call is answered or not, and it times out after 10 seconds. After three unsuccessful password attempts (each timeout counts as 1), Asterisk hangs up. Stated another way, you have about 30 seconds to enter your password after the phone first rings. Then Asterisk disconnects the call. To enter your password, key in the touchtone numbers which match the numerical password code you specified in your [callout] context (see below). Then press the pound (#) key. Note that a web page will not display at this web address unless you enter the portion of the address following the question mark. Nor will a call be placed unless the sip/bv/ syntax precedes a phone number. We did this for security reasons.

Before the above script will work, you also need to add the following context to the bottom of the extensions_custom.conf configuration file discussed above. Make sure you change the password 24681234 to something very secure. After all, it’s your phone bill! Once you make this change, it won’t take effect until you restart Asterisk. The easiest way to do that is to access setup within AMP, click Incoming Calls, then click the Submit Changes button, then click on the red bar which appears. Count to 10 and your changes should be operational.

[callout]
exten => s,1,Authenticate(24681234)
exten => s,2,DISA(no-password|from-internal)


One Ringy-Dingy. As we mentioned at the beginning of this article, there may be times when you don’t have access to a web browser and the cost of completed outgoing calls is astronomical. Or you may just find it more convenient to place a quick call to your Asterisk server rather than firing up a web browser. For the One Ringy-Dingy option to work, you must place a call to your Asterisk server from a phone that can accept incoming calls directly (not most hotel rooms or pay phones), and you must call from a phone with a legitimate CallerID number. Ideally, for this option to work in an unattended way all the time, you’d want to have a separate Direct-Inward-Dial (DID) number dedicated to this task. Why? Because, once Asterisk detects one ring on this number, it will issue a Congestion tone (fast busy) and immediately build and then process a DISA script to call you back. That obviously isn’t a desirable response on your regular phone number. We still will configure DISA to prompt for a password when the return call is answered, but callers may be a little surprised if they call you at home, hear a fast busy, and then immediately get a return call from your home asking them for a password.

Here are the steps to get this set up. We’ll tweak our Asterisk@Home/AMP dial-in context to turn on support for tracking incoming calls by DID. Then, for security, we’ll build a separate context for this DID number to isolate it from our default AutoAttendant which manages the rest of our incoming calls. Once we have the DID context created, we’ll build a DID entry in AMP to support this incoming line. Next, we’ll drop in the code to actually process the incoming call and build and execute the scripts necessary to make the return call. There are several new and very important Asterisk features that we’ll be taking advantage of. First, we’ll be executing code based upon a ringing phone line as opposed to an answered call. Second, we’ll show you how to execute context code and scripts after a call ends. And finally, we’ll be setting a maximum call limit on the outgoing DISA calls just to provide some food for thought on how you can better harrass your teenagers. So let’s get started.

Tweaking the Dial-In Context. From our previous articles, you will recall that Asterisk@Home and other Asterisk systems using the Asterisk Management Portal (AMP), rely upon the [from-sip-external] context in extensions.conf to process incoming calls. Right now, we have that context pointed to our AutoAttendant context which we built in the extensions_custom.conf file. You can read all about how to build the AutoAttendant here. The AutoAttendant implementation effectively disabled support for AMP’s DID Routes, but it secured your Asterisk system by reducing the number of points of attack to one, i.e. every incoming call had to flow through our one AutoAttendant. For those just getting started with Asterisk, this was a good thing. But, now that you’re an expert, we need a little more flexibility because we want to set up a DID line just to handle requests for DISA services, and we don’t want incoming calls on that line going to our AutoAttendant. Why? Because, with a separate DISA DID, we can eliminate any costs in placing calls to the Asterisk server requesting remote dial tone. How? This DID will never be answered. All it will do is ring once or twice before handing out a Congestion tone, and that activity will be sufficient to capture the incoming call’s CallerID and then set in motion the DISA return call process … hence the name One Ringy-Dingy. Won’t the telemarketers be thrilled! Just think of this DID as a toll-free number without having to pay for a toll-free number. To turn on support for AMP’s DID Route management, just add the following line to your [from-sip-external] context, and be sure you add it immediately below the [from-sip-external] label. See how easy this is when there’s some documentation (HINT!).

include => ext-did


Configuring an AMP DID Route. Now that we’ve activated DID Route support for Asterisk, let’s actually build a DID Route to show you how it’s done. This presupposes that you’ve ordered an additional DID from one of your providers and that you already have a trunk for that provider set up. When you order the additional DID, make sure that you specify that you do not want voicemail activated on this DID. Or, if you have control of the voicemail setup for this DID, turn it off. The reason is that, when incoming calls to this number get a congestion tone from Asterisk, that will activate the voicemail option with most providers. That, in turn, defeats our purpose of not answering calls ringing on this DID line to save money.

To create the DISA DID route, use a web browser to access the Asterisk Management Portal. Then choose AMP->Setup->DID Routes. Plug in the DID phone number you wish to assign to DISA duty. For the Destination, click on the Custom App button, and then enter the following: custom-teliax-in,8435551212,1 substituting your actual DID number. Click the Submit button and then the red bar to reset Asterisk. Note that AMP does no error checking for this custom context other than looking for the word "custom." Remember, we haven’t even built this context yet! But let’s do it now.

DISA Custom Contexts. For our One Ringy Dingy example, we’re going to use two different providers. The DID line is rented from Teliax.com. That’s where you call to trigger a callback. But Teliax is just too expensive for actual outgoing or incoming calls so we’re using two TelaSIP trunks (with permission): one to return the call to the original caller (that’s you or me) and one to place our outgoing DISA call (to the callee). If you happen to use TelaSIP, remember that you don’t need two separate accounts for the outbound calls since TelaSIP gives you two voice paths with your single line account.

There are three separate contexts we need to create at the bottom of extensions_custom.conf file to make all of this work. Keep in mind that there are three steps in the One Ringy-Dingy process: (1) you place a call to your DID number, and your Asterisk server detects the incoming call; (2) your Asterisk server calls you back, and you’re given dialtone after successfully entering your DISA password; and (3) your Asterisk server lets you to place an outgoing DISA call for a specified length of time to anywhere permitted in this context’s custom DISA dialplan. Here are the three contexts to support the three functions: (1) [custom-teliax-in] listens to the DID line for an incoming call and then sets up and executes the code setting the return call in motion; (2) [custom-telasip-callout] actually manages the return call once someone answers and authenticates the user for DISA service; and (3) [custom-telasip] sets the dialplan and timeout options for the call and then actually connects and times the call if the caller satisfies the dialplan rules. If you’re a little confused at this point, an example may help. Skip down to the Free Samples section below and actually try a test call to see how all this works. Then come back, and we’ll finish building it for your own Asterisk system.

Incoming Call Context. So a user first calls your DID number to initiate a DISA callback. Let’s start with the code that handles the incoming call on your DID line:

[custom-teliax-in]
exten => 8436541010,1,NoOp(Incoming call from Teliax #8436541010)
exten => 8436541010,2,Congestion
exten => 8436541010,3,Hangup
exten => h,1,SetCIDNum(${CALLERIDNUM:2})
exten => h,2,System(echo channel: SIP/telasip-gw/${CALLERIDNUM} > /tmp/${CALLERIDNUM})
exten => h,3,System(echo context: custom-telasip-callout >> /tmp/${CALLERIDNUM})
exten => h,4,System(echo extension: ${CALLERIDNUM} >> /tmp/${CALLERIDNUM})
exten => h,5,System(echo priority: 1 >> /tmp/${CALLERIDNUM})
exten => h,6,System(echo callerid: 2025560000 >> /tmp/${CALLERIDNUM}) ; Your CallerID for your TelaSIP account goes here
exten => h,7,System(echo sleep 30 > /tmp/${CALLERIDNUM}.2)
exten => h,8,System(echo cp /tmp/${CALLERIDNUM} /var/spool/asterisk/outgoing >> /tmp/${CALLERIDNUM}.2)
exten => h,9,System(chmod 775 /tmp/${CALLERIDNUM}.2)
exten => h,10,System(/tmp/${CALLERIDNUM}.2)
exten => h,11,Hangup()

Let’s walk through each line of the script so that you’ll know what needs to be changed on your own system if you use this. The first three lines get processed every time someone calls your dedicated DISA DID number. Adjust the numbers in each line to match your DISA line’s 10-digit CallerID number. Line 1 just outputs a message to the console indicating that there’s an incoming call on this line. Line 2 sends the Congestion tone back to the incoming caller. Line 3 is just a safety net to be sure the call is hung up as far as Asterisk is concerned. The next 11 lines beginning with exten=>h tell Asterisk what to do when it detects that the caller has hung up on the call. This code gets processed after the caller hangs up or after Asterisk processes the Hangup command in the third line.

Line h,1 is very important. Different providers relay CallerID numbers in different ways. Teliax sends a U.S. number as +14041234567. Many providers (including BroadVoice and TelaSIP) just send the ten-digit number for incoming U.S. calls. You need to know this because we’ll need to format the CallerID number properly to match what your outgoing provider expects to see when you process the return call in step 2. So how do you know what the CALLERIDNUM format is? Run asterisk -r from a command window and watch the information about an incoming call on the DID line you plan to use. If the format is a 10-digit number, then change SetCIDNum(${CALLERIDNUM:2}) to SetCIDNum(${CALLERIDNUM:0}) which means "use every digit as received." If the number is in the format 1+8901234567 then leave the expression the way it is. It basically says strip off the first two digits and store the rest as the CallerID number.

Line h,2 is where you identify which of your VoIP providers will be used to place the return call in step 2. If you’re using TelaSIP (as we are here), then just make sure the outbound trunk name matches your entry for this provider under AMP->Setup->Trunks. If you’re using an IAX provider instead of a SIP provider, change SIP to IAX2 as well. Finally, be sure what is now a 10-digit CallerID number is formatted properly for the return call through your dialback provider. TelaSIP wants a 10-digit number. GoIAX, for example, expects to receive a 1 and then the area code and number. For GoIAX, just insert a 1 before the CallerID number.

Line h,3 identifies the context which will actually place the return call. Line h,6 is where you specify the CallerID number of your Asterisk trunk that will be placing the return call. Teliax, for example, lets you spoof the CallerID for your calls so this is where you would enter the main number of the White House (if that’s your thing): 2024561414. Finally, in line h,7 you can set how much of a delay will be imposed before your Asterisk system places the return call. It’s set to 30, but you can change it to meet your requirements. The rest of this code should work as is.

Callback Context. Now we’re ready for Asterisk to place the return call. The previous code actually sets the call in motion with a 30 second delay and then a call to the CallerID number specified using the [custom-telasip-callout] context. But here’s the code that actually manages the callback. It also prompts for the DISA password once someone answers. Line s,3 is where you set the DISA password. Make it secure! Remember: IT’S YOUR PHONE BILL!

[custom-telasip-callout]
exten => s,1,Background(silence/2)
exten => s,2,Background(asterisk-friend)
exten => s,3,Authenticate(6373)
exten => s,4,Background(pls-wait-connect-call)
exten => s,5,DISA(no-password|custom-telasip)

When someone answers the return phone call, Asterisk counts to 2 and then says, "Asterisk is your friend. Please enter your password and press the pound key." That’s the person’s clue to enter their DISA password. Three guesses and Asterisk hangs up. If the password is successfully entered, Asterisk provides dialtone and passes the call to the [custom-telasip] context in s,5.

DISA Outbound Call Context. Once dialtone is provided, the user can enter whatever digits are permitted in the [custom-telasip] context’s dialplan. Here’s ours:

[custom-telasip]
exten => _1NXXNXXXXXX,1,AbsoluteTimeout(600)
exten => _1NXXNXXXXXX,2,Macro(dialout-trunk,8,${EXTEN:1},)
exten => _1NXXNXXXXXX,3,Macro(outisbusy) ; No available circuits
exten => T,1,Playback(thank-you-for-calling)
exten => T,2,Playback(goodbye)
exten => T,3,Hangup

Lines 1, 2, and 3 specify that the user has one and only one dialing option: dial 1 and then a 3-digit U.S. area code and then a 7-digit U.S. phone number. Line 1 is where you set the maximum duration for the call in seconds (600=10 minutes). Line 2 is where you really have to be careful. It has three gotcha’s. First, you need to identify which trunk will be used to place the DISA call. If your provider only offers one dial path per circuit, then this trunk cannot be the same one as what’s specified in custom-telasip-in,h,2 above or you’ll get an "all circuits are busy" message. In short, you need one outbound trunk for the callback and another outbound trunk to place the actual DISA call from dialtone. If you use TelaSIP’s residential plan, then you can use the same trunk for both. Second, you need to know the number of the dialout trunk to be used for the DISA call. AMP numbers all of your outbound trunks. If you look in the [globals] context at the top of the extensions_additional.conf file, you’ll see all of your outbound trunks labeled as OUT_1, OUT_2, etc. Find the number of the one that matches the name of the outbound trunk you want to use for your DISA call. Replace the "8″ in line 2 of [custom-telasip] with the appropriate number you wrote down. Don’t use OUT_8, just 8. Third, you’ve got to get the DISA number dialed by the user properly formatted for the provider that will be handling the call. In our case, we accepted only numbers beginning with 1 plus a U.S. area code plus a 7-digit number, but TelaSIP doesn’t want the 1 so we strip it off. If, however, your provider (such as GoIAX) expects a 10-digit number with a leading 1, then you’d adjust line 2 above to look like this: exten => _1NXXNXXXXXX,2,Macro(dialout-trunk,8,${EXTEN},). This tells Asterisk not to strip off the first digit before sending the rest to dialout-trunk 8. The extension entries beginning with an upper case T tell Asterisk what to do when the maximum duration of a call expires.

That about does it. To complete your work, save the three new contexts to extensions_custom.conf and then restart Asterisk.

Making a Test Call. To be sure everything is working as it should, start up Asterisk in interactive mode using the Command Line Interface (CLI) so that you can actually watch what’s happening when calls are placed and received. This works best if you connect to your Asterisk server through SSH from a Mac or PC. SSH comes with every Mac and the syntax is simple: ssh root@AsteriskIPaddress. If you’re still chained to Microsoft, download Putty from the Mother Country, and you can do the same thing using a Windows machine. Once you’re logged in as root, issue the following command: asterisk -r. Quit ends your Asterisk CLI session, and exit logs you out of your SSH session. Now issue the command: set verbose 10 to get maximum information. Then place a call to your DISA DID number from your cellphone and watch what happens. You should see the call being processed without being answered. Asterisk will then issue a Congested tone and disconnect. Your CLI display will remain quiet for 20 seconds, and then the return call will be placed. When you answer the call, you’ll be prompted for your password. Enter it, wait for dial tone, dial a number with 1, then area code, then 7-digit number and watch what happens. If the call fails, exit from CLI by typing Quit. Change to the /tmp directory: cd /tmp. Do a directory listing in reverse date order: ls -all -t -r. At the bottom of the list should be two files with the area code and number of your cellphone. One will have a .2 extension. Display the contents of the file without the extension: cat 6781234567 using your cellphone number. The contents should look something like this. If not, check your typing in the contexts we added and try again. Keep in mind that editing either of these two files is pointless. They both get built on the fly in the h section of the [custom-teliax-in] context depending upon the CallerID of the caller. Linux will automatically handle deletion of files from the /tmp directory in due course.

[root@asterisk1 tmp]# cat 6781234567
channel: SIP/telasip-gw/16781234567
context: custom-telasip-callout
extension: 6781234567
priority: 1
callerid: 2025560000

And, yes, we know the extension specified does not match the extension (s) actually entered in the [custom-telasip-callout] context. But, fear not, your call will fall back to the s extension when Asterisk can’t find an entry matching your cell phone number. Why did we do this? Well, you may want to have Asterisk do something special when it knows it’s you calling from your cellphone or one of your kids calling from your house at the lake. For example, you might want to provide a wide open dialplan for yourself with a different password as well. Remember, people can spoof your CallerID number so make the password VERY SECURE! Here’s how it would look. Substitute your own cellphone number and a new password. Once you make the changes, restart Asterisk and then place another call from your cellphone to your DISA DID and try entering a local extension or a foreign phone number if your regular dialplan supports it. Enjoy!

[custom-telasip-callout]
exten => 6781234567,1,Background(silence/2)
exten => 6781234567,2,Background(asterisk-friend)
exten => 6781234567,3,Authenticate(63738488537)
exten => 6781234567,4,Background(pls-wait-connect-call)
exten => 6781234567,5,DISA(no-password|from-internal)

exten => s,1,Background(silence/2)
exten => s,2,Background(asterisk-friend)
exten => s,3,Authenticate(6373)
exten => s,4,Background(pls-wait-connect-call)
exten => s,5,DISA(no-password|custom-telasip)


Free Samples. Everybody loves free samples so here’s one for you. You may remember Gene Willingham and the good folks at TelaSIP, the winner and least expensive provider in our unlimited U.S. long distance calling shootout. In a moment of weakness, TelaSIP’s agreed to let you try out our One Ringy-Dingy service (known affectionately around our house as the One Rinky-Dink Service because of the amount of time we’ve wasted on this). Anyway, it uses our Asterisk server and TelaSIP’s long distance bandwidth. Calls are limited to 10 minutes after which the callee will hear "Goodbye" followed by a click. You’ll get two beeps followed by a fast busy. In other words, time’s up! Here’s how to use this free service. Call from a phone in the U.S. with CallerID. Call the number shown on the map (inset) which is one of our DIDs in Charleston, South Carolina. The call will never be answered so you won’t be billed for the long distance call. Within 10 seconds, you should hear a fast busy. Just hang up, and our Asterisk server will call you back within 30 seconds. Listen carefully! You’ll be provided a random password for your call, and then you’ll be prompted to key it in. With your phone keypad, just do that and press the pound key (#). If you get a message that it’s incorrect, just try again (HINT: I told you to listen carefully). We used to have a fixed password, but the war-dialers were abusing the system so now it’s random. Back to the drawing boards, boys! Once you successfully enter the password, you’ll then get a DISA dial tone. Dial 1 and then the area code and phone number of someone you love (in the United States only). This is an excellent way for you to check out the voice quality of TelaSIP calls without spending a dime. Just don’t abuse the offer or this paragraph may magically disappear … as will the free calls. For those that don’t know us, we don’t record your calls, and we don’t store the number of the person you’re calling although the Asterisk logs probably have it for a while anyway. For security purposes, we do log your CallerID and the time of your call just in case you do something you shouldn’t be doing and the FBI traces the call back to us. If any of the above offends you, exercise your constitutional right to not use this free service. And a final reminder: none of this works if you don’t have CallerID enabled when you call. How would we know where to call you back? You won’t believe how many calls we receive with a CallerID of Asterisk. That obviously won’t work either.

Homework. There’s really a fourth DISA option. Under this scenario, you would call your home number, have the AutoAttendant answer, and then press 4 for DISA. After providing your password, Asterisk would hang up and call you back with DISA dial tone using the same type scripts we implemented in One Ringy Dingy above. The advantage of this approach is you don’t need a separate DID line to support DISA. The disadvantage is you have to pay for a one minute phone call to your home number each time you want remote dial tone. But it would save the expense of lengthy calls to your home just to use your outbound trunks. We’ll leave it to you to figure this one out. It shouldn’t take you long now that you understand how all the pieces fit together.

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