Monthly Archives: November 2005

Want a rock-solid PBX at a rock-bottom price: free! Gosh, you haven't heard that since our column a few weeks ago introducing Asterisk® 1.2. What a difference two weeks makes. The final version of Asterisk@Home 2.0 was released the day before Thanksgiving and, from the looks of things, it's darn near perfect! You not only get the latest version of Asterisk (version 1.2), you also get the latest and greatest version of Linux, CentOS 4.2; the latest Festival Speech Engine (1.96); the latest version of the Asterisk Management Panel (1.10.010); the Flash Operator Panel (version 0.24); Digium® card auto-configuration; fax support; loads of AGI scripts including weather forecasts and wakeup calls; xPL support; and the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support. And it all still fits on a single CD!

NOTE: Version 2.1 was posted late Wednesday, November 30. Our new 2.1 tutorial will be available here on Friday, December 2.

The installation process is pretty straightforward. You download an ISO image from here, burn a CD (click here if you need a refresher course), use an old clunker PC or an under $200 WalMart special (see inset), insert the CD you made, plug your machine into the Internet and turn it on. Then watch while Asterisk@Home loads CentOS/4.2 and all the Asterisk and Linux goodies imaginable: Apache, SendMail, Comedian Mail, SugarCRM, MySQL, PHP, phpMyAdmin, SSH, Bluetooth, the Asterisk Management Panel, the Flash Operator Panel, Call Detail Reporting, and on and on. We've covered how to use most of the Linux products in our Mac HOW-TO's (see sidebar), and they work exactly the same way with Asterisk@Home so keep reading. And, yes, this install will reformat (aka ERASE) your hard disk before it begins, but it now warns you first.

Loading CentOS/4 and Asterisk 1.20. Here's how the 2.0 install went for us, and we'll walk you through the few very minor issues that still remain to be manually tweaked. Once the install begins, you can expect to eat up about 25 minutes with the CentOS 4.2 install. The install CD then will eject itself, reboot the system, and begin the Asterisk compile and installation. That takes about 25 more minutes to complete.

Securing Your Passwords. When it's finished and reboots, log in as root with password as your password. Type help-aah for a listing of the passwords that need to be changed. Change them all NOW!

passwd
passwd admin
passwd-maint
passwd-amp
passwd-meetme


Getting the Latest CentOS Updates. Once your system is secure, load all of the application updates for CentOS 4.2. There are about forty of them as we write this so be patient. The update command to issue is yum -y update.

Activating Bluetooth Support. Once the updates are completed, activate Bluetooth support if you plan to use it with our Follow-Me Phoning proximity detection application. Run setup, down arrow to System Services, press ENTER, down arrow to bluetooth and press the space bar, tab to OK, press ENTER, tab twice to Quit and press ENTER.

Rebuilding Zaptel. First, reboot your system: shutdown -r now. Because a new version of the kernel is installed as part of the update, you'll need to rebuild support for ZAP devices. Log in as root and type rebuild_zaptel. Reboot once more and you're all set to go: shutdown -r now.

Simplifying SSH. If you're going to be connecting to other servers from your new Asterisk@Home 2 system using SSH or SCP, then build your new RSA key pair now. This lets you use SSH and SCP (secure copy) without having to enter a password each time. You can also automate backups and proximity detection scripts as we've explained previously here. 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 other Asterisk servers, Linux machines, and Macs. There's probably a way on PCs as well, but I've given up on that platform particularly after Sony's latest security stunt so you're on your own there. From your Asterisk2 server using SCP, the command should look like the following (except use the private IP address of each of your other Asterisk or Linux servers instead of 192.168.0.104). Provide the root password to your other servers (one at a time) when prompted to do so.

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

On a Mac running Mac OS X, the command would look like this (using your username and your Mac's IP address, of course):

For user access only: scp /root/.ssh/id_rsa.pub wardmundy@192.168.0.104:/Users/wardmundy/.ssh/authorized_keys
For full root access: scp /root/.ssh/id_rsa.pub root@192.168.0.104:/var/root/.ssh/authorized_keys


Once the file has been copied to each server, try to log in to your other server from your Asterisk 2 Server with the following command using the correct destination 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 other server by typing exit.

Installing WebMin. We don't build Linux systems without installing WebMin, the Swiss Army knife of the Linux World. You can use it to start and stop services, check logs, adjust startup scripts, manage cron jobs, babysit your SendMail server, and many, many other tasks that are downright painful without it. If you ever need help from others, WebMin is a great tool for letting others help you.

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 used to be 10000 to 20000 so there's a conflict on port 10000 with WebMin. Beta 6 fixed this, but the final version doesn't have the change. So, if it still says 10000 on your system, 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. Then restart Asterisk: amportal restart. Finally, to stop WebMin when you're finished using it, issue this command: /etc/webmin/stop. You can start it any time you need it, and then use a web browser to access it. But there's no need to consume processing resources running a second web server when you're not using it.

Basic System Configuration. To get a basic Asterisk system up and running, you only need to do a few things. First, you need an Outbound Trunk to actually deliver your outbound calls to Plain Old Telephones (POTS). Second, you need to configure an Outbound Route to tell Asterisk which trunk to use to deliver your outbound calls to the intended recipients. Third, you need to configure at least one extension so that you can plug in some sort of telephone instrument to place and receive calls using your new Asterisk server. The phone can be a hardware device such as an IP telephone or a POTS phone, or it can be a software device such as a free IP softphone. The advantage of IP telephones and softphones is that they require no additional hardware besides your Asterisk server. A POTS phone or our favorite, a 5.8GHz wireless phone system with up to 10 extensions, both require an additional piece of hardware although some of the newer IP wireless phones give you the best of all worlds (see inset). To use a POTS phone, the hardware required is either a circuit board with an FXS port or an external black box (ATA device) such as a Sipura SPA-1001. If you also want to connect your Ma Bell phone line to your Asterisk server, then you need a circuit board with an FXO port or an external black box (ATA device) such as a Sipura SPA-3000. Our favorite is the SPA-3000 because it has both FXO and FXS ports in a box the size of a pack of cigarettes for under $100.

Setting Up An Outbound Trunk. You configure an outbound trunk using your web browser and the Asterisk Management Portal (AMP). But first, you have to have an account with a service provider. This is the company that carries your calls from your Asterisk server to plain old phones in your neighbor's house or Aunt Betty's in California. With VoIP, it's a good idea to have two providers, but today let's start with one. We'll save you some time and lots of money. Unless you make substantial international calls regularly, use TelaSIP/VoipExpress. If you want to know why, read the full article here. Or just try a free call for yourself using our server. Basically, $5.95 a month gets you a local number in your choice of area code with free incoming calls, and 2¢ per minute for outbound calls to anywhere in the U.S. $9.95 a month buys you all of that plus free outbound calls in the area code of the phone number you select. $14.95 a month gets you a number in the area code of your choice with unlimited incoming calls and unlimited outbound calls to anywhere in the U.S. There are no sneaky add-on fees and no obnoxious terms of service. Just be sure to tell them to configure your account for use with Asterisk. The also have very reasonable business plans. If, on the other hand, you'd prefer to try another provider, take a look at our easy setup guides for most of the major VoIP providers here.

Once you have your account name and password, point your web browser to the IP address of your new Asterisk 2.0 server and log in as maint with the password you selected. Then choose AMP->Setup->Trunks->Add SIP Trunk assuming you're using TelaSIP. NOTE to existing users: if you already have an Asterisk server using your TelaSIP account, don't use the same account at the same time on your new Asterisk@Home 2.0 server! Plug in the CallerID number you were assigned for your account. Set Maximum Channels to 2. For the Dial Rules, use the following (substituting your local area code for 404 below):

1|NXXNXXXXXX
NXXNXXXXXX
404+NXXXXXX

In the Outgoing Settings section, name your trunk telasip-gw. Then enter the following for the Peer Details using your own account name for username and fromuser and using your own assigned password for secret:

context=telasip-in
dtmfmode=rfc2833
fromuser=youraccountname
host=gw3.telasip.com
insecure=very
secret=yourpassword
type=peer
username=youraccountname

Leave the Incoming Settings section blank, and in the Registration String, enter the following using your account name and password:

youraccountname:yourpassword@gw3.telasip.com

Click the Submit Changes button, and then click the red bar to reload Asterisk. Now we need to add the context which will actually process the incoming calls from TelaSIP. Choose AMP->Maintenance->Config Edit->extensions_custom.conf and add the following code at the bottom of the file substituting your new phone number for 4041234567. Save the file and reload Asterisk to finish the setup.

[telasip-in]
exten => 4041234567,1,NoOp(Incoming call on TelaSIP #4041234567)
exten => 4041234567,2,Dial(local/200@from-internal,20,m)
exten => 4041234567,3,VoiceMail(200@default)
exten => 4041234567,4,Hangup


Configuring an Outbound Route. Now we need to tell Asterisk where to send our outbound calls when we dial them. To get started, we'll just send everything to the TelaSIP trunk we just configured. Choose AMP->Setup->Outbound Routing->Add Route. For Route Name, use Outside. Leave the password blank. For Dial Patterns, enter the following:

NXXXXXX
NXXNXXXXXX
1NXXNXXXXXX

For the Trunk Sequence, choose SIP->telasip-gw from the drop-down list. Then click Submit Changes and then click the red bar to save your Outbound Routing setup.

Configuring an Extension. You have to have an extension to make and receive calls with Asterisk@Home so let's build one. Choose AMP->Setup->Extensions->SIP to begin. For the Extension Number, let's use 200 to keep things simple. For the Display Name, make up something that tells where this phone will be located, e.g. Kitchen. For the Outbound CID, use 200. For secret, make up a password for this extension. For Voicemail and Directory, choose Enabled. Plug in your password again. Type in your email address, and, if you want to also be paged when you get a new voicemail, type in a pager email address. Click the Yes button beside Email Attachment, and leave the other settings alone. Now click the Submit button. You'll see a couple of ugly error messages. Ignore them. It's a beta bug. Just click the red bar to save your changes and reload Asterisk.

Downloading a Free Softphone to Test Asterisk. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator. Both are free! Just install and then configure with the IP address of your Asterisk@Home 2 server. For username and password, use your extension number and password from above. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best value in the marketplace with excellent build quality and feature set is the under $100 GrandStream GXP-2000. It has support for four lines, speaks CallerID numbers, has a lighted display, and can be configured for autoanswer with a great speakerphone. Short of paying three times as much, that's as good as desktop phones get. If you want to use Asterisk throughout your home, buy a good 5.8GHz wireless phone system with plenty of extensions (our two favorites are shown in the insets below) and then purchase an SPA-3000 to connect up both your home phone line and all your cordless phones. Our tutorial will show you how. The final option is to use a wireless IP phone which is the best of both worlds, a cordless phone that talks IP telephony without an ATA blackbox such as the Uniden UIP1868 (see also insets above).

Activating Email Delivery of VoiceMail Messages. When you're out and someone leaves you a voicemail message, Asterisk@Home will let you forward that voicemail message to your email address as a .wav file which can be played within most email client software. Or you can have Asterisk@Home send an instant message to your cell phone or pager telling you who called, what their phone number was, and how long a voicemail message the person left for you. Or you can do both. In addition, you can tell Asterisk@Home whether to delete the voicemail from your Asterisk server after sending it to your email account. In short, you now can manage all of your incoming email and voicemail from a single place, your email client. In order to send out emails from your Asterisk@Home server, you'll need to make two changes. First, make this adjustment to the /etc/hosts file on the server. Since anonymous emails are blocked by most ISPs, you'll need a fully-qualified domain name for your server. The easiest one to use is the fully-qualified domain name that your ISP assigns to the IP address for your broadband connection. Don't forget to update it when your ISP changes your IP address. To find out what your fully-qualified domain name is, go to a command prompt on your Asterisk server and type: nslookup 123.456.789.001 substituting your public IP address for the preceding numbers. Then write down the name entry without the trailing period. Now edit the hosts file: nano /etc/hosts. Move the cursor to the line which begins 127.0.0.1, and then move the cursor over the first letter of the first domain name shown, usually asterisk1.local. Now type in the fully-qualified domain name you previously wrote down and add a space after your entry. Don't erase the existing entries! Save your settings: Ctrl-X, y, enter. Now restart network services on your Asterisk machine: service network restart. Second, go into AMP->Maintenance->Config Edit->vm_general.inc with a web browser. Change the serveremail entry to an email name at the fully qualified domain you used in your /etc/hosts file above. Then save your configuration and restart Asterisk. If you continue with this setup and still don't receive emails, here's another configuration change that is sometimes necessary. On the Asterisk terminal, log in as root. Switch to the directory where the SendMail configuration file is stored: cd /etc/mail. Make a backup of the config file: cp sendmail.cf sendmail.cf.bak. Then issue the following command: echo CGasterisk.dyndns.org >> sendmail.cf. Substitute the actual domain name of your Asterisk server for asterisk.dyndns.org, but be sure it's preceded by CG with no intervening spaces.Then reboot your server and try again: shutdown -r now.

To configure the voice mail forwarding options, go into the Setup tab of the Asterisk Management Portal using a web browser. Click on Extensions and then click on an extension you already have configured. In the Voicemail and Directory section of the form, enter either (or both) your email address and your pager or cellphone's text messaging address. To email the voicemails as attachments, just click Yes beside Email Attachment. To delete the voicemail message from your voicemail inbox after sending it to your email address (not recommended until you first get it working correctly), click Yes beside Delete Vmail. If you want to further customize the email message which is sent, just edit vm_email.inc from AMP's Maintenance->Config Edit screen using your favorite web browser. For those using a dynamic IP address with phones at remote locations connecting to your Asterisk server, we'll show you how to automate the process of changing your Asterisk server's IP address in a future column.

Fixing Call Recording. This link explains the process as well as we could. After making the two changes, call recording inbound and outbound works reliably.

Fixing Paging. If you want to use paging with your Asterisk system, you'll need to perform a little magic to get it working with your full duplex sound card in Asterisk@Home 2.o. For the step-by-step, review this posting on SourceForge.

Fixing Directory Lookup. Usually, pressing the pound key (#) from any phone connected to your Asterisk server calls up a directory lookup function using the Asterisk Management Portal (AMP); however, Digium renamed one of the voice prompts in the 1.2 release of Asterisk which broke this function in AMP. If you simply log into your server as root and issue the following command, it will create a symbolic link to the renamed file and will permanently fix the problem:

ln -s /var/lib/asterisk/sounds/dir-intro-fn.gsm /var/lib/asterisk/sounds/dir-intro-oper.gsm


Managing Incoming Calls. For long time readers of this column, you already know that our recommended strategy for handling incoming calls is to set up a simple Stealth AutoAttendant. Basically, this is a welcome message that greets your callers and then transfers them to an extension or ring group of your choice. The advantage of this approach is that it also lets callers like you press buttons to navigate through various options on your Asterisk system without advertising them to the public at large. If you're just getting started with Asterisk, you can read all about setting up a Stealth AutoAttendant here. If you'd prefer to manage your incoming calls with AMP, you'll still need to fix the [from-sip-external] context in the extensions.conf file, or all your incoming SIP and IAX calls will ring busy. To fix it, choose AMP->Maintenance->Config Edit->extension.conf->from-sip-external. Comment out all the lines in the existing file by adding a semicolon at the beginning of each line. Then add the following line, save your changes, and reload Asterisk.

exten => _.,1,Goto(from-pstn-timecheck,s,1)


Where To Go From Here. Once you've got a functioning Asterisk system, you're ready to move on to the really cool things that make Asterisk a one-of-a-kind PBX. There are customized weather reports, web and phone-based dialers from a MySQL address book, incoming fax to PDF conversion with email delivery, blacklisting of telemarketers, bluetooth proximity detection so that your home or office calls automatically transfer to your cellphone when you depart with your bluetooth device, and on and on. You'll also want to take a more in-depth look at many of the topics we've covered above. For a complete catalog of all of our Asterisk projects and everything else we've written about Asterisk@Home, go here. Then take a look at a terrific writeup from the other side of the globe: Asterisk@Home for Dumb-Me. Finally, there's an Asterisk@Home Handbook Wiki project under development that's worth a careful look. Enjoy!

Want a rock-solid PBX at a rock-bottom price: free! It’s been over a year since the initial release of Asterisk®, and this week the new stable 1.2 release finally hit the street. If you’re just dying to try it and can’t wait for Asterisk@Home to catch up so that you’ll have all your favorite goodies to go with Asterisk, here’s the quick solution for you. First, download and install the latest Asterisk@Home 2.0 beta. This may not work with Asterisk@Home versions below 2.0! See the Comments to today’s article before you try it. The drill is pretty simple. You download an ISO image from here, burn a CD (click here if you need a refresher course), use an old clunker PC or a shiny new WalMart special (see inset for the unbelievable price!), insert the CD, plug your machine into the Internet and turn it on. Then watch while Asterisk@Home loads CentOS/4 and all the Asterisk and Linux goodies you’ll ever need: Apache, SendMail, Comedian Mail, SugarCRM, MySQL, PHP, phpMyAdmin, SSH, and on and on. We’ve covered how to use most of these products in our Mac HOW-TO’s (see sidebar), and they work exactly the same way with Linux so keep reading. And, yes, this install will reformat (aka ERASE) your hard disk before it begins. Once it’s finished, change all the default passwords by logging in to your new Asterisk@Home server as root with password as your password, and type help-aah for a list of the passwords that need to be changed, or go here for our complete security tutorial. A list of new features in Asterisk 1.2 is available here.

Editor’s Note: This version of Asterisk has been superceded. For the latest tutorial on or after February 1, click here.

When you finish the Asterisk@Home 2.0 beta install, we’ll first get the latest updates for CentOS/4. Then we’ll load the new Asterisk 1.2 stable release. Here’s how. Log in to your new Asterisk server as root, or better yet, use SSH to log in as root, and then cut and paste each command below in order:

amportal stop
yum -y update

amportal start


Checking Your Install. The Asterisk@Home install takes a little less than an hour, and the Asterisk 1.2 upgrade will set you back another 30 minutes or so. Not bad for free! Once Asterisk restarts, you should be able to log in to your Asterisk Management Portal by pointing a web browser at the IP address of your Asterisk system. Now choose AMP->Maintenance->Asterisk Info and make sure everything is up an running. The Version block should display Asterisk 1.2.0 with the time that you completed the build. If you’ve already got an IP phone or if you’d like to try a free IP-based softphone with your PC, go here next. Last but not least, you need a phone number and service provider so make this link your last stop, and you’ll be off to the races. Enjoy!

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. Then you can skip around to your heart’s content.

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.