Category Archives: Telephony

Want a rock-solid PBX at a rock-bottom price: free! Yep, it's been a week and here we go again! Asterisk@Home 2.2 has hit the street primarily because of a bug-fix release of Asterisk®! Now you get the latest version of Asterisk (version 1.2.1), and 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 Portal (1.10.010); the Flash Operator Panel (version 0.24); Open A2Billing; Digium® card auto-configuration; fax support; loads of AGI scripts including weather forecasts and wakeup calls; xPL support; the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support; plus some more bug fixes. And it amazingly still fits on a single CD!

Editor's Note: This version of Asterisk@Home has been superceded. For the latest tutorial on or after March 13, click here.

The installation process is pretty straightforward. You download the 2.2 ISO image from here, burn a CD (click here if you need a refresher course), use an old clunker PC or a $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 Portal, 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 1.2.1. Here's how the 2.2 install went for us, and we'll walk you through getting everything set up so that it can be used as a production server. 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 five 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 unless you have Digium cards. If you do, read the caution which follows before proceeding. There are about forty of them as we write this so be patient. The update command to issue is yum -y update.

Dec. 27 WARNING: There is an additional wrinkle with running yum update with Asterisk@Home 2.2 if you have Digium cards. Read this SourceForge thread and make sure you understand what's necessary to put Humpty back together again before running yum update if you have Digium cards in your Asterisk system.

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 yum update, you'll need to rebuild support for ZAP devices. Log in as root and type the following commands:

cd /usr/src/zaptel
make install-udev
rebuild_zaptel


Reboot your system: shutdown -r now. Now log in as root again and type genzaptelconf. Reboot once more and you're all set to go: shutdown -r now. You only need to rebuild Zaptel when there is a kernel update as there was with this yum update.

Simplifying SSH. If you're going to be connecting to other servers from your new Asterisk@Home 2.2 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 2.2 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 Asterisk@Home 2.2 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@Home 2.2 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. Note that there is a new version of WebMin since our article on Asterisk@Home 2.0. It fixed a major security flaw in some of WebMin's perl scripts so you'll definitely want this upgrade:

cd /root
mkdir webmin
cd webmin
wget http://internap.dl.sourceforge.net/sourceforge/webadmin/webmin-1.250-1.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. We really don't want the WebMin server starting up each time the OS reboots so do the following. Once you're logged in to WebMin, choose System->Bootup and Shutdown and then click on webmin. Click the No button beside Start at boot time, and then click the Save button. 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 version 2.2 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. They 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@Home 2.2 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 2.2 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. See the Comments to this post for a more versatile approach which will let you use your TelaSIP line for Ring Groups.

[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. Be sure you also delete the sample outbound route that came with the install, or your outbound calls may go nowhere. Finally, click the red bar to save your new 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 2.2 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. If you don't have your own domain, the easiest alternative is to use 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 second line which reads 127.0.0.1 asterisk1.local , 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 entry! Save your settings: Ctrl-X, y, enter. Now restart network services on your Asterisk machine: service network restart. Next, you need to modify the email message which delivers your voicemails so that it includes your fully-qualified domain name. Don't do this in AMP, or you'll mess up the formatting of the email message. You can download a fresh copy here if you need it. Instead, use nano: nano -w /etc/asterisk/vm_email.inc. Press Ctrl-W, type /cgi, and press the enter key. You're now positioned where you need to type either the fully-qualified domain name for your Asterisk server or the private IP address if you only want to read your emails from behind your firewall. When you start typing, the text display is going to jump all over the place because of word wrap. Don't freak out. You haven't messed anything up. Once you complete your entry, don't erase or change anything else. Save the file: Ctrl-X,Y, then enter. Now 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. Finally, if your ISP doesn't permit downstream mail servers (that's you), then take a look at this link which will show you how to designate your ISP as your SMTP smart host using SendMail.

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. 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.

Call Recording Fixed. This update fixes inbound and outbound call recording which now works reliably. You can set your preferences for call recording when you set up each extension. The recordings are stored in /var/lib/asterisk/monitor unless you set other preferences in agents.conf.

Paging Fixed. If you want to use paging with your Asterisk system, the sound card problems in Asterisk@Home 2.1 and Asterisk 1.2 have been resolved. Review this posting on SourceForge for additional details. It now works with full and half-duplex sound cards. Thanks, Tracy!

Wakeup Calls Broken. Asterisk@Home 2.2 includes part of a new version of the Wakeup Call program. Unfortunately, it doesn't work. Fortunately, the old version still does. And, we've also fixed the code in the new version. To reload the old version, log into your server as root and issue the following commands:

cd /var/lib/asterisk/agi-bin
mv wakeup.php wakeup-new.php
wget http://nerdvittles.com/wakeup.zip
unzip wakeup.zip
rm wakeup.zip
chmod 775 wakeup.php
chown asterisk:asterisk wakeup.php

Or, if you'd prefer to use a patched copy of the new version, log into your server as root and issue these commands instead:

cd /var/lib/asterisk/agi-bin
mv wakeup.php wakeup-new.php
wget http://nerdvittles.com/wakeupnew.zip
unzip wakeupnew.zip
rm wakeupnew.zip
chmod 775 wakeup.php
chown asterisk:asterisk wakeup.php

There are basically two problems with the new wakeup code. The first is that a number of the voice files are assumed to be in /var/lib/asterisk/sounds/extra, a directory which doesn't exist in the Asterisk@Home setup. We've fixed that problem in the wakeupnew.zip file which you can install above. The second problem is that the wakeconfirm application is missing entirely. This is used to prompt the user to answer a question ... to make sure the person is REALLY awake. This application is not enabled in the default setup so just don't activate it until we can do a little more investigative work.

Adjusting Call Parking Extensions. Traditionally, dialing extension 700 parked a call on Asterisk@Home systems, and the call could be picked up by dialing the parked extension in the range of 701 to 799. To retain this setup, you'll need adjust the settings in features.conf. Go to AMP->Maintenance->Config Edit->features.conf and make it look like this:

[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-799 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in


Directory Lookup Fixed. Pressing the pound key (#) from any phone connected to your Asterisk server now calls up a directory lookup function using the Asterisk Management Portal (AMP).

Max Channels Bug. A bug has been reported because of a deprecated command that makes Asterisk@Home's calculation of maximum channels invalid. To fix it, goto AMP->Maintenance->Config Edit->extensions.conf->macro-dialout-trunk and comment out line s,7 so that it looks like this:

;exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})

Then insert a new line s,7 just below it which looks like this:

exten => s,7,GotoIf($[ ${GROUP_COUNT()} > ${OUTMAXCHANS_${ARG1} ]?108)

Then click the Update button and reload Asterisk to activate the change.

Tweaking SIP.conf. There are a few changes we recommend you make in the [general] context of the sip.conf file. Using the Asterisk Management Portal, go to AMP->Maintenance->Config Edit->sip.conf. It's a good idea to include your actual CallerID number of your default outbound trunk here instead of Default. We also recommend that you add allow=gsm just below the existing allow=alaw line in the file. You may also want to include progressinband=yes which assures that callers will hear ring tones when placing calls even if your provider doesn't provide them. This is a fairly common complaint with BroadVoice in particular. Don't forget to reload Asterisk once you make these changes: AMP->Setup->Incoming Calls->Submit Changes and then click the Red bar.

Connecting Remote Extensions or a Remote Asterisk Server. If you plan to connect remote extensions to your Asterisk server, then add the following entries to sip.conf above using your own fully-qualified domain name and the first three octets of the private IP address of your Asterisk server:

externip = myasteriskbox.dyndns.org
localnet=192.168.0.0/255.255.255.0
nat=yes

You'll also either need to define your Asterisk server as a DMZ device in your firewall setup, or you can open the following UDP ports and map all of them to the private IP address of your Asterisk box: 4569, 5004-5082, and 10000-20000. If you only hear half of a conversation with a remote extension, it's usually a NAT problem meaning you probably forgot to do the port mapping drill. We plan to cover remote extensions and interconnecting Asterisk servers in more detail in a future column so stay tuned. In the meantime, if you have a dynamic DNS connection that changes your IP address regularly and you don't want to wrestle with manually updating your Asterisk server each time there's a change, just download chandave's sip reload script and copy it to your /etc/asterisk directory. Then execute the following commands while logged in as root:

chown asterisk:asterisk /etc/asterisk/sip_reload.sh
export EDITOR=nano
crontab -e
00,05,10,15,20,25,30,35,40,45,50,55 * * * * root /bin/sh /etc/asterisk/sip_reload.sh >/dev/null 2>&1

Save your crontab addition in the usual nano way, and you're all set: Ctrl-X, y, then enter. This clever little script will make sure your Asterisk server always knows its own IP address regardless of how often your ISP changes it. And you'll never lose inbound connectivity from remote extensions or servers for over 5 minutes. If you'd like to read the full discussion, visit this link on the Voxilla forum.

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->extensions.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)


New Custom Speed Dialing. Asterisk@Home 2.2 has a built-in speed dialing utility. The reserved speed dial numbers are 300 to 399. Adding a number to your speed dial list is easy. Just pick up an extension and dial 300-3xx-6781234567 where 3xx is the speed dial code you want to create and 6781234567 is the phone number you want dialed when you enter the speed dial code. Just make sure you enter the number to be called in a format that is supported by your Asterisk dialplan, i.e. if outside calls need to be preceded by a 1 or a 9, then the number should be entered in a matching format. You can look up speed dial numbers by dialing an asterisk followed by the 3-digit speed dial code, e.g. *301 would tell you the number stored in speed dial 301. If you need additional flexibility with both web browser and phone access as well as 1 to 5-digit speed dial codes, download our free AsteriDex robodialer.

Fixed A2Billing: Asterisk Calling Card Platform. This web-based application allows you to generate and issue calling cards to individuals so that they can place calls remotely through your Asterisk server. If you've always wanted to be just like AT&T, here's your Big Chance! There's very little that you can do with an AT&T calling card that can't be done as well or better by you using A2Billing. And, it won't take an M.B.A. to undercut AT&T's calling card rates and still make buckets of money. All you need now are a few customers. Heck, I'll sign up with you. I sign up for everything. But first, a word of caution. Assuming your Asterisk server has web exposure on the Internet, you need to secure the admin and root passwords in this application whether you use it or not. To access the application, go to http://192.168.0.104/a2billing/ using the actual internal IP address of your Asterisk server. Log in as root with a password of myroot. Click on the ADMINISTRATOR tab in the left column and then click Show Administrator. Now click on the Edit button beside each of the two administrator accounts and change the passwords to something secure. If you really would like to learn more about it, documentation for the application is available here. And, if you decide to use the application, you'll need to uncomment the actual dialplan lines in extensions_custom.conf and reload Asterisk:

; CallingCard application
; un-comment the 5 lines below to use this app
;exten => _X.,1,Answer
;exten => _X.,2,Wait,2
;exten => _X.,3,DeadAGI,a2billing.php
;exten => _X.,4,Wait,2
;exten => _X.,5,Hangup

Footnote: The missing A2Billing code from Asterisk@Home 2.1 has been added. You can read all about the problem here. There's also a pretty good step-by-step setup guide for Asterisk@Home 2.2 here.

SugarCRM Contact Management. Asterisk@Home includes the latest and greatest version of the best open source contact management application on the planet, SugarCRM. You access the application with a web browser: http://192.168.0.104/crm/ substituting the private IP address of your Asterisk box, of course. Specify admin for your username and password for your password. Whether you use the application or not, change the admin password. It's easy. Just click the Administrator link under Welcome admin. Then click the Change Password button. Complete documentation for the application is available here. If contact management is your thing, knock yourself out, and we'll talk to you next spring when you finish getting everything set up to run your business. It's a great product, but be prepared to invest lots of time in the project if you expect to use it productively.

Other Out-of-the-Box Utilities. Asterisk@Home 2.2 comes bundled with a number of additional utilities. Here are some of them. You can retrieve the current time by dialing *60. If the time is wrong, you can reset your default time zone by logging into your server as root and typing config. A current weather report for New York is available by dialing *61. You can change the city by following our previous tutorial which is available here. To set up a wakeup call from any extension, dial *62. To determine the phone number of any extension, just dial *65. You can use the default MeetMe conferencing system from any or all of your extensions by dialing 8 plus the number of an existing extension. Additional conference rooms can be added by editing meetme_additional.conf. Finally, you can record customized voice prompts for your system by dialing 5678 from any extension. Before this will work, edit the extensions_custom.conf file (AMP->Maintenance->Config Edit->extensions_custom.conf) and uncomment the seven lines shown below which are located at the bottom of the file. Just remove the leading semicolons. You'll also need to uncomment the following line near the top of file at the beginning of the [from-internal-custom] context: ;include => custom-recordme.

;[custom-recordme]
;exten => 5678,1,Wait(2)
;exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
;exten => 5678,3,Wait(2)
;exten => 5678,4,Playback(/tmp/asterisk-recording)
;exten => 5678,5,Wait(2)
;exten => 5678,6,Hangup

Once you make a recording, it needs to be moved to /var/lib/asterisk/sounds/custom with a new filename.gsm, e.g. mv /tmp/asterisk-recording.gsm /var/lib/asterisk/sounds/custom/hihoney.gsm. Then change the ownership of the file: chown asterisk:asterisk /var/lib/asterisk/sounds/custom/hihoney.gsm. You then can play the recording with a line like this in your dialplan: exten=>s,1,Playback(custom/hihoney) where hihoney is the name you assigned to the recording without its .gsm extension.

Where To Go From Here. After you get 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!

Some Recent Nerd Vittles Articles of Interest...

One of the first things you learn after you purchase a new aquarium is that you're much better off with two tanks rather than one. This generally doesn't make the little Mrs. too happy but, after you explain that each dead fish represents a $100 bill, most wives usually come around. YMMV! You'll learn much the same thing with Asterisk®. It's not that new versions are that terrible. They're not. In fact, they're generally a quantum leap better and less buggy than the version you're replacing. It's just that programmers have this death wish to always "improve" everything. I've had lots of programmers work with me over the years, and the hardest thing to get across to them short of branding or tattooing the message directly onto their heads is this: "If It Ain't Broke, Don't Fix It!" The gotcha in Asterisk 1.2 was leaving out the sound card driver, chan_oss.so, which broke every company's paging system ... at least all those companies that now depend upon Asterisk. Digium® apparently has fixed the problem in the just released Asterisk 1.2.1, but it still ranks up there as a big mistake that should and could have been avoided. The silver lining in this tale is that it prompted a single individual, Tracy Carlton, to stay up an entire night trying to figure out what had come unglued. In the process, Tracy uncovered why the chann_oss.so driver has never worked with half-duplex sound cards, you know, the type that comes standard in virtually every PC on the planet. Guess what? Now it works with one simple tweak. You can read all about sound cards and paging in Tracy's post on SourceForge. Don't miss it! And, speaking of not missing it, the next time you come through our hometown of Atlanta, don't miss the new Georgia Aquarium (see insets). Known affectionately as Fish Depot because it was funded privately by Bernie Marcus of Home Depot fame, it's truly one of a kind and destined to become one of this country's national treasures. The Showstopper: an unbelievably friendly, 3,000 pound Beluga whale (inset below). And the mascot: a little orange fish called a garibaldi appropriately named ... Deepo.

What does all this have to do with Backups and Redundancy? Plenty. You'll save yourself and your customers a boatload of grief by thoroughly testing new Asterisk releases before you put them into production. Asterisk is such an exciting technology that it's always tempting to deploy the latest, greatest version immediately to take advantage of some new bell or whistle. The downside is what happened with paging in Asterisk 1.2. I'm no longer dumbfounded by developers that don't try everything before releasing an upgrade. There just isn't time, and usually it's not very exciting work either. Plus, it requires a substantial hardware investment. Your best defense is a sandbox (aka clunker) PC that can be used for all sorts of things. First, you can try out new releases. And second, for about $100 on Ebay, you can buy a great Asterisk@Home PC which can do all sorts of other things, too ... like automatic full system backups. If you're feeling rich, buy another Wal-Mart special for under $200, and you've got a great alternate server/backup/sandbox all rolled into one. Or, if you previously built our dual-system Bluetooth Proximity Detection System, then you're all set. And, yes, we're going to get software RAID 2 working with Asterisk@Home 2 one of these days for those that want a mirrored hard disk, but first things first. RAID doesn't offer the same protection as a good, frequent backup. Mirrored RAID drives immediately copy everything from one disk to another, mistakes and all. So, if you clobber a config file or driver by pressing the wrong button, a mirrored drive won't help you at all. It'll just give you a second copy of your original mess. So today we're going to show you a painless backup strategy that'll get you through almost any disaster except perhaps Katrina or a fire that totally destroys your home or office.

Overview. Our project for today is to build a bullet-proof backup system between your two Asterisk@Home systems. Yes, Asterisk@Home can be configured to make backups, but it puts them on the same hard disk as your system, not too helpful with a disk crash. That's not to say you shouldn't implement the Asterisk@Home backups. They're a great resource when you accidentally clobber something. It's easy: AMP->Setup->Backup and Restore->Add Backup Schedule. Give it a name and set everything to Yes. Then select Run Daily at Midnight. Click Submit Changes and then the red bar to reload Asterisk. Just don't forget to clean out some of the daily backups from time to time. They live in /var/lib/asterisk/backups under a directory name matching whatever you named your backup schedule.

The dual-PC backup system we're going to build today works well for almost any two Linux systems since Asterisk@Home really is a Linux system. It also works swimmingly on systems running BSD or MacOS X. If you need help getting either or both of your Asterisk systems set up, start here. And the software cost of the project: still $0. For those with a small business and one or more satellite offices, next week we'll tackle tying multiple Asterisk systems together for instant communications between your offices. Yet another freebie for those that deploy today's backup system. But we're getting ahead of ourselves. The drill for today is to use our previous SSH automatic login trick to enable us to do scheduled backups in the background on one or both of your systems. Lucky for us, CentOS includes the rsync utility which makes mirror image backups using SSH as the transport between two systems. Here's the step-by-step if you want to follow along.

Password-Free SSH Access. As with our Proximity Detection System, we want to use SSH for our backups to protect our sensitive data. After all, there's no reason that your two Asterisk@Home systems have to be in the same building. One can be across town or around the globe. This backup system will work exactly the same ... just slower. But to automate the process, we need to use crontab to schedule the backups. That means there will be no user interaction in the backup process. As you know, SSH typically prompts for a password when you connect to a remote resource. So here's the trick we introduced in our Bluetooth project. We'll do the same thing for backups. And, if you want to backup both of your systems, you'll need to perform this trick in both directions, i.e. from Server A to Server B and from Server B to Server A. For ease of explanation, we're going to refer to the servers as Asterisk Main and Asterisk Backup. You can extrapolate for your own systems from there. We're going to go through this example assuming you want to backup the data on Asterisk Main to Asterisk Backup. Once you complete the steps, there's no reason you can't reverse the process and also backup Asterisk Backup to Asterisk Main. We'll leave that for you to work out on your own. Just let us know if you get stumped.

To begin, log in to Asterisk Main as root. Then, 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 Asterisk Backup server from Asterisk Main using SCP. The command should look like the following (except use the private IP address of your Asterisk Backup server instead of 192.168.0.104). Provide the root password to Asterisk Backup 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, you now should be able to log in to Asterisk Backup from Asterisk Main with SSH without being prompted for a password. Let's try it. Log in to Asterisk Backup from Asterisk Main with the following command using the actual IP address of Asterisk Backup, 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. While you're logged in to Asterisk Backup, let's create the directory where your Asterisk Main system will be stored: mkdir /backup. We're assuming you have sufficient disk space on Asterisk Backup to store the complete contents of Asterisk Main. If you're not sure, issue the df command on both systems and be sure. Here's a sample from our system:

[root@asterisk1 backup]# df

Filesystem 1K-blocks Used  Available Use% Mounted on
/dev/hda2  27980732 4625788 21933596 18% /
/dev/hda1   101086   11077  84790    12% /boot
none         62612       0  62612     0% /dev/shm


The device we care about is /dev/hda2. That's where your backup data will reside. The Available column will tell you how much space is left on Asterisk Backup. When you run the same df command on Asterisk Main, the Used column for /dev/hda2 will tell you how much room you need to reserve on Asterisk Backup to hold the backup. Finally, if you should ever want to delete the /backup directory and all the backup data stored on Asterisk Backup, here's the command to do it: rm -rfd /backup. You get no warnings! Now log out of Asterisk Backup by typing exit. Then rerun the df utility on Asterisk Main to see how much space will be needed for the backup. Incidentally, all of the remaining steps in this tutorial will be performed on Asterisk Main.

Using rsync for Asterisk@Home Backups. There are a number of great tutorials on rsync. If you want to learn more than what's covered below, start here. We did. The actual command we're going to use to make our backups is shown below. It will make the first backup and then freshen it each time you run it in the future. We recommend running the command manually the first time to be sure everything is working as it should. When the rsync session completes (5-10 minutes on a new system with a 100 megabit Ethernet connection between the servers), issue the following command to check the rsync log for errors: tail /var/log/backup.log. If errors are reported, you can open the log with nano (nano /var/log/backup.log) to search for errors (Ctrl-W, failed to open). Then close nano (Ctrl-X, n). Typically, errors involve changes during the backup process, such as deleting a voicemail message while the backup is in process. Normally, running rsync again will solve any problems. Once you get a clean backup, move on to the Automating Backups step below.

rsync -avvz --delete --exclude "/proc/" --exclude "/mnt/" --exclude "[cC]ache" -e ssh / root@192.168.0.104:/backup/ > /var/log/backup.log

WARNING: If you cut-and-paste the code above, make sure you adjust the quotation marks which get mangled by the WordPress blog.
Both Linux and Asterisk like and want regular quotation marks in all commands or disaster awaits!

The only change you'll need to make in the command above is to replace the quotes and the 192.168.0.104 IP address with the actual IP address of your Asterisk Backup server. This will backup everything on Asterisk Main except cache files, the /mnt directory (which includes the contents of any mounted CD-ROM), and the /proc directory (which includes a bunch of stuff that would be unchanged if you did a clean Asterisk@Home install).

Restoring Your Asterisk@Home System. For future reference, to do a full system restore of Asterisk Main with a new hard disk, you would first reinstall Asterisk@Home from the same installation disk used to create Asterisk Main. Then you would restore the backup data from Asterisk Backup by logging into Asterisk Main as root and manually issuing the following commands:

cd /
rsync -avv -e ssh root@192.18.0.104:/backup/ . > /var/log/restore.log

Of course, we're assuming the IP address of your Asterisk Backup server is 192.168.0.104 in the above example so change it to properly reflect your Asterisk Backup's IP address. And don't forget the space and period and space in the above command. That's what tells rsync to restore your backup to the current folder location.

Automating Backups. Now that you have rsync working reliably, we want to automate the Linux backup process so that it runs each night without user intervention. This will assure that you never lose more than one day's data which usually isn't life threatening in the telephone business. If it is, you can adjust the crontab schedule to better meet your requirements. To begin, download the backup script here. Now log in as root and copy the script to /etc/asterisk on your Asterisk Main server. Reset the permissions: chmod 775 /etc/asterisk/backup.sh. Fix the ownership: chown asterisk:asterisk /etc/asterisk/backup.sh. And adjust the IP address in the file to match the IP address of your Asterisk Backup server: nano -w /etc/asterisk/backup.sh. Finally, add the script to your crontab to kick off at 2:01 a.m. each morning, save your update (Ctrl-X, Y, then enter) and you're all set to go.

export EDITOR=nano
crontab -e
01 02 * * * /etc/asterisk/backup.sh >/dev/null 2>&1


Coming Attractions. Yes, we're finally going to cover faxing later this week or early next week. And, the #1 requested feature for Asterisk, call presence monitoring with pickup button support, is now a reality. And soon we'll show you how to use it to mimic the key telephone instruments that corporate America just can't seem to live without. There also are new versions of both Asterisk (1.2.1) and Asterisk@Home (2.2) with more bug fixes so we'll have an updated HOW-TO guide for you shortly. Finally, as mentioned previously, we'll connect up our two new Asterisk servers to communicate with each other and start sharing extensions and dialplans as soon as Weekly UPDATE-itis comes to a close.

BroadVoice in the News. Well, it looks like our old pals at BroadVoice are back in the news ... at least with the Better Business Bureau. See this link where BroadVoice finishes in second place for most complaints out of 77 telephone providers over a three year period. And, better yet, BroadVoice would retain or perhaps better their ranking if the numbers were based upon just the past 12 months. In fact, all but one of the complaints filed against BroadVoice were filed within the past 12 months. And, yes, there is a silver lining to this story, too. BroadVoice for the most part gives BBB complaints the same great treatment that we've received when inquiring about a problem ... nada!

We’ve pretty well documented how you can set up Bluetooth Proximity Detection using a bluetooth headset or cellphone with your Asterisk® PBX. Once configured, phone calls in your home or office can automatically be transferred to your cellphone whenever you take off carrying your bluetooth device. In our original articles, you’ll recall that we recommended a bluetooth headset as the ideal way to track your comings and goings at very little cost. But today, we want to add another bit of magic to the project and also give you something to tell Santa about. It’s the incredible Tom Tom Go, a portable GPS device that has the Garmin’s of the world shaking in their boots because this thing is so easy to use and just does everything right. Having endured absolutely terrible built-in GPS units in both Cadillac and Mercedes Benz automobiles and not-much-better Garmin units, take it from us. Buy a TomTom. Our Cadillac GPS had to be replaced four times and finally with a unit from a later 2005 Escalade before you could store a location and call it up without crashing the entire system. And GM wonders why they’re losing money. Worse yet, to put a name with a location using the Mercedes GPS still requires a trip to the manual. It’s that painful and unintuitive! So, when your friendly car dealer touts the built-in GPS devices in their automobiles, JUST SAY NO! We haven’t seen a built-in unit yet that doesn’t suck.

Some Hints for GPS manufacturers: Nobody wants a GPS that reproduces an entire paper street map on a 4 inch screen. We’re trying to figure out how to get somewhere! What’s important is the name of the street you’re on, the names of the next few cross streets, and how far to drive until the next turn. Which way to turn with a little advance warning is also a nice touch. And, by the way, we’re smart enough to know not to be fiddling with the GPS while the car is moving so don’t lock the damn unit when the car is moving. In case you haven’t heard, some cars can actually have more than one person riding in them at the same time. They’re called passengers, and they can even chew gum and operate a GPS while seated in the passenger seat. Bozos! Here’s the best hint of all: Go buy a TomTom for you and your company and copy what they’ve done.

TomTom at a Glance. With the TomTom, you can either get the flash drive model 300 with the entire U.S. and Canada maps on a single chip, or there’s a hard disk version 700 which also gives you a hands-free speakerphone and phonebook for use with your compatible bluetooth cellphone. Treo 650 fans are SOL. Both TomTom models provide automatic route calculation and turn-by-turn directions through a built-in speaker. And, with either unit, you also can get traffic reports and the latest weather forecasts not to mention points of interest alerts showing where every "safety" camera is located in many European countries. If you can’t figure out how to use a TomTom in under 15 minutes, you need to stay away from anything that uses electricity. Yes, it’s that good. And all the TomTom units are Linux-based so you can download the source code and build your own GPS if that’s your thing. Circuit City will even let you try a TomTom for two weeks and return it for a full refund. So take them up on the offer. Then, if you decide to buy one, take the Circuit City unit back and buy it on the web. It’ll save you over $200! With the current $50 rebate and free U.S. traffic reports, the TomTom Go 300 can be purchased for under $500 with some careful shopping (HINT: PriceGrabber.com). See how nice we are! We could have encouraged you to click on the link below and actually make us a little money … but who needs it, right?

Using a TomTom for Proximity Detection. Once you have your TomTom Go device, you also can use it in our Bluetooth Proximity Detection system in lieu of a headset. Here’s how. Because the TomTom unit is designed to allow you to download weather reports and traffic information using your bluetooth-enabled cellphone, that, of course, means the TomTom unit talks bluetooth. So, just like your bluetooth headset, the only trick is discovering the MAC address of the TomTom’s built-in bluetooth adapter. The device is designed to operate on its internal battery for a day at a time. Thus it’s pretty simple to carry the unit to your Asterisk server and turn it on. Once it’s on, tap the screen once, tap the right arrow icon twice to move to the third page, and then tap TomTom Weather. When prompted whether to set up your wireless internet connection now, tap Yes. While logged into your Asterisk server, type hcitool scan and, presto, your TomTom unit will dutifully report its MAC address for all the world to see:

[root@asterisk1 tmp]# hcitool scan
Scanning ...
00:0F:3D:4B:DF:E0 n/a

N/A doesn’t tell you much, but it’s your TomTom. Trust us on this one. Once you have this tidbit of information, simply edit your ruhome script and plug in the required information:

mainasteriskbox=192.168.0.118
deviceuser=TomTom
devicemac=00:0F:3D:4B:DF:E0


Now, when you drive your automobile into your garage, your home phones will come back to life. The only wrinkle, of course, is that you’ll need to leave your GPS unit powered on while you’re home. Otherwise, powering down the TomTom would tell your Asterisk server that you had departed again. Yeah, you’re right. It’s not ideal, but it did give us the opportunity to offer a great tip for your Christmas wish list. And it ought to get you thinking that this particular device is well-suited to integrate into your home automation system to turn on the lights and hot tub. With home automation system software such as Indigo and its AppleScript object model and dictionary or Salling Clicker, you don’t have to worry about the TomTom turning itself off in the garage because all we really need is the proximity "trigger" to alert Indigo to turn on the lights. Once on, you can program Indigo to define how long the lights stay on before automatically turning themselves off again. Problem solved.

Implementing Proximity Detection on a Single Asterisk@Home 2.1 Server. Our original articles on how to deploy a Bluetooth Proximity Detection System assumed you were using two Asterisk servers, one for phone calls and a second version 2 server for proximity detection. However, now that Asterisk@Home 2.1 is soup, we thought it would be helpful to show you how to run the entire system using a single Asterisk@Home 2.1 server. First, download the updated proximity detection software here. Once you unzip the file, you’ll note that there’s a new ruhome2 file. The only changes you’ll need to make from the original tutorial are to substitute the ruhome2 file for the original ruhome file and to copy homecheck.agi to /var/lib/asterisk/agi-bin on your Asterisk@Home 2.1 server. Don’t forget to reset the file permissions as previously explained. Once you make these two simple changes, the entire proximity detection system can be run from your one and only Asterisk@Home 2.1 server.

Manually Managing In and Out Status with a SIP Phone. We’ve also received several queries from readers asking for a simple way to turn off the proximity detection system and to manually manage your IN or OUT status using buttons on a SIP telephone. In other words, when you leave your home or office, you want to press a button on the phone to tell your Asterisk server whether you’re IN or OUT. Yes, you can do it on a per extension basis using *72, but the proximity detection system transfers all calls based upon the location of your bluetooth device. To do the same thing manually, first remove the ruhome application from your crontab by logging into your server as root and deleting that line (Ctrl-K) from your crontab file. Then save your changes: Ctrl-X, Y, Enter.

export EDITOR=nano
crontab -e

Then add the following code to the [from-internal-custom] context of your extensions_custom.conf file and reload Asterisk. Note that, in the code below, you’ll have to change the name of the file in the /tmp directory from WARD to whatever filename you’re currently using with your proximity detection system. This is the deviceuser variable in your ruhome script. You’ll also need to modify the permissions on this file after logging into your Asterisk server as root, or this won’t work: chmod 666 /tmp/WARD.

exten => 46,1,Answer ; IN to deactivate call forwarding
exten => 46,2,Wait(1)
exten => 46,3,System(cp -f /var/lib/asterisk/agi-bin/notnull.file /tmp/WARD)
exten => 46,4,Playback(call-forwarding)
exten => 46,5,Playback(de-activated)
exten => 46,6,Wait(1)
exten => 46,7,Playback(goodbye)
exten => 46,8,Hangup

exten => 688,1,Answer ; OUT to activate call forwarding
exten => 688,2,Wait(1)
exten => 688,3,System(cp -f /var/lib/asterisk/agi-bin/null.file /tmp/WARD)
exten => 688,4,Playback(call-forwarding)
exten => 688,5,Playback(activated)
exten => 688,6,Wait(1)
exten => 688,7,Playback(goodbye)
exten => 688,8,Hangup

Once you make these changes, you can pick up any extension and dial IN (46) when you’re IN or OUT (688) when you’re away. You can also assign these "extensions" to buttons on almost any SIP telephone instrument if you want one-touch dialing.

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. There’s also an index of all the previous articles which you can review here.

Want a rock-solid PBX at a rock-bottom price: free! Gosh, you haven't heard that since our column three days ago introducing Asterisk@Home 2.0. You didn't know this was going to be a bi-weekly event, did you? Well, neither did we, but we're happy to oblige when a new release hits the street because it always has something great tucked away inside. Asterisk@Home 2.1 was released two days ago and, from the looks of things, it's another winner! 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 Portal (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; the SugarCRM Contact Management System with the Cisco XML Services interface and Click-to-Dial support; plus some bug fixes. And there are a few new surprises that we'll get to shortly. Best of all, it amazingly still fits on a single CD!

Editor's Note: This version of Asterisk@Home has been superceded. For the latest tutorial, click here and scroll down the page.

The installation process is pretty straightforward. You download the 2.1 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 Portal, 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 2.1 install went for us, and we'll walk you through getting everything set up so that it can be used as a production server. 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 five 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. Then reboot. Now log in as root again and type genzaptelconf. Reboot once more and you're all set to go: shutdown -r now. You only need to rebuild Zaptel when there is a kernel update as there was with this yum update.

Simplifying SSH. If you're going to be connecting to other servers from your new Asterisk@Home 2.1 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 2.1 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 Asterisk@Home 2.1 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@Home 2.1 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. Note that there is a new version of WebMin since our article on Asterisk@Home 2.0 earlier this week. It fixed a major security flaw in some of WebMin's perl scripts so you'll definitely want this upgrade:

cd /root
mkdir webmin
cd webmin
wget http://internap.dl.sourceforge.net/sourceforge/webadmin/webmin-1.250-1.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. We really don't want the WebMin server starting up each time the OS reboots so do the following. Once you're logged in to WebMin, choose System->Bootup and Shutdown and then click on webmin. Click the No button beside Start at boot time, and then click the Save button. 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 version 2.1 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. They 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. Be sure you also delete the sample outbound route that came with the install, or your outbound calls may go nowhere. Finally, click the red bar to save your new 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. If you don't have your own domain, the easiest alternative is to use 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 second line which reads 127.0.0.1 asterisk1.local , 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 entry! Save your settings: Ctrl-X, y, enter. Now restart network services on your Asterisk machine: service network restart. Next, you need to modify the email message which delivers your voicemails so that it includes your fully-qualified domain name. Don't do this in AMP, or you'll mess up the formatting of the email message. You can download a fresh copy here if you need it. Instead, use nano: nano -w /etc/asterisk/vm_email.inc. Press Ctrl-W, type /cgi, and press the enter key. You're now positioned where you need to type either the fully-qualified domain name for your Asterisk server or the private IP address if you only want to read your emails from behind your firewall. When you start typing, the text display is going to jump all over the place because of word wrap. Don't freak out. You haven't messed anything up. Once you complete your entry, don't erase or change anything else. Save the file: Ctrl-X,Y, then enter. Now 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. 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.

Call Recording. This update fixes inbound and outbound call recording which now works reliably. You can set your preferences for call recording when you set up each extension. The recordings are stored in /var/lib/asterisk/monitor unless you set other preferences in agents.conf.

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 sound card in Asterisk@Home 2.1. For the step-by-step, review this posting on SourceForge. It now works with full and half-duplex sound cards. Thanks, Tracy!

Directory Lookup. Pressing the pound key (#) from any phone connected to your Asterisk server now calls up a directory lookup function using the Asterisk Management Portal (AMP).

Tweaking SIP.conf. There are a few changes we recommend you make in the [general] context of the sip.conf file. Using the Asterisk Management Portal, go to AMP->Maintenance->Config Edit->sip.conf. It's a good idea to include your actual CallerID number of your default outbound trunk here instead of Default. We also recommend that you add allow=gsm just below the existing allow=alaw line in the file. You may also want to include progressinband=yes which assures that callers will hear ring tones when placing calls even if your provider doesn't provide them. This is a fairly common complaint with BroadVoice in particular. Don't forget to reload Asterisk once you make these changes: AMP->Setup->Incoming Calls->Submit Changes and then click the Red bar.

Connecting Remote Extensions or a Remote Asterisk Server. If you plan to connect remote extensions to your Asterisk server, then add the following entries to sip.conf above using your own fully-qualified domain name and the first three octets of the private IP address of your Asterisk server:

externip = myasteriskbox.dyndns.org
localnet=192.168.0.0/255.255.255.0
nat=yes

You'll also either need to define your Asterisk server as a DMZ device in your firewall setup, or you can open the following UDP ports and map all of them to the private IP address of your Asterisk box: 4569, 5004-5082, and 10000-20000. If you only hear half of a conversation with a remote extension, it's usually a NAT problem meaning you probably forgot to do the port mapping drill. We plan to cover remote extensions and interconnecting Asterisk servers in more detail in a future column so stay tuned. In the meantime, if you have a dynamic DNS connection that changes your IP address regularly and you don't want to wrestle with manually updating your Asterisk server each time there's a change, just download chandave's sip reload script and copy it to your /etc/asterisk directory. Then execute the following commands while logged in as root:

chown asterisk:asterisk /etc/asterisk/sip_reload.sh
export EDITOR=nano
crontab -e
00,05,10,15,20,25,30,35,40,45,50,55 * * * * root /bin/sh /etc/asterisk/sip_reload.sh >/dev/null 2>&1

Save your crontab addition in the usual nano way, and you're all set: Ctrl-X, y, then enter. This clever little script will make sure your Asterisk server always knows its own IP address regardless of how often your ISP changes it. And you'll never lose inbound connectivity from remote extensions or servers for over 5 minutes. If you'd like to read the full discussion, visit this link on the Voxilla forum.

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->extensions.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)


Custom Speed Dialing. Asterisk@Home 2.1 now has a built-in speed dialing utility. The reserved speed dial numbers are 300 to 399. Adding a number to your speed dial list is easy. Just pick up an extension and dial 300-3xx-6781234567 where 3xx is the speed dial code you want to create and 6781234567 is the phone number you want dialed when you enter the speed dial code. Just make sure you enter the number to be called in a format that is supported by your Asterisk dialplan, i.e. if outside calls need to be preceded by a 1 or a 9, then the number should be entered in a matching format. You can look up speed dial numbers by dialing an asterisk followed by the 3-digit speed dial code, e.g. *301 would tell you the number stored in speed dial 301. If you need additional flexibility with both web browser and phone access as well as 1 to 5-digit speed dial codes, download our free AsteriDex robodialer.

A2Billing: Asterisk Calling Card Platform. This web-based application allows you to generate and issue calling cards to individuals so that they can place calls remotely through your Asterisk server. If you've always wanted to be just like AT&T, here's your Big Chance! There's very little that you can do with an AT&T calling card that can't be done as well or better by you using A2Billing. And, it won't take an M.B.A. to undercut AT&T's calling card rates and still make buckets of money. All you need now are a few customers. Heck, I'll sign up with you. I sign up for everything. But first, a word of caution. Assuming your Asterisk server has web exposure on the Internet, you need to secure the admin and root passwords in this application whether you use it or not. To access the application, go to http://192.168.0.104/a2billing/ using the actual internal IP address of your Asterisk server. Log in as root with a password of myroot. Click on the ADMINISTRATOR tab in the left column and then click Show Administrator. Now click on the Edit button beside each of the two administrator accounts and change the passwords to something secure. If you really would like to learn more about it, documentation for the application is available here. And, if you decide to use the application, you'll need to uncomment the actual dialplan lines in extensions_custom.conf and reload Asterisk:

; CallingCard application
; un-comment the 5 lines below to use this app
;exten => _X.,1,Answer
;exten => _X.,2,Wait,2
;exten => _X.,3,DeadAGI,a2billing.php
;exten => _X.,4,Wait,2
;exten => _X.,5,Hangup

Footnote: There's also a little missing A2Billing code which needs to be added. You can read all about it here. There's also a pretty good step-by-step setup guide for Asterisk@Home 2.1 here.

SugarCRM Contact Management. Asterisk@Home includes the latest and greatest version of the best open source contact management application on the planet, SugarCRM. You access the application with a web browser: http://192.168.0.104/crm/ substituting the private IP address of your Asterisk box, of course. Specify admin for your username and password for your password. Whether you use the application or not, change the admin password. It's easy. Just click the Administrator link under Welcome admin. Then click the Change Password button. Complete documentation for the application is available here. If contact management is your thing, knock yourself out, and we'll talk to you next spring when you finish getting everything set up to run your business. It's a great product, but be prepared to invest lots of time in the project if you expect to use it productively.

Other Out-of-the-Box Utilities. Asterisk@Home 2.1 comes bundled with a number of additional utilities. Here are some of them. You can retrieve the current time by dialing *60. If the time is wrong, you can reset your default time zone by logging into your server as root and typing config. A current weather report for New York is available by dialing *61. You can change the city by following our previous tutorial which is available here. To set up a wakeup call from any extension, dial *62. To determine the phone number of any extension, just dial *65. You can use the default MeetMe conferencing system from any or all of your extensions by dialing 8400. Additional conference rooms can be added by editing meetme_additional.conf. Finally, you can record customized voice prompts for your system by dialing 5678 from any extension. Before this will work, edit the extensions_custom.conf file (AMP->Maintenance->Config Edit->extensions_custom.conf) and uncomment the seven lines shown below which are located at the bottom of the file. Just remove the leading semicolons. You'll also need to uncomment the following line near the top of file at the beginning of the [from-internal-custom] context: ;include => custom-recordme.

;[custom-recordme]
;exten => 5678,1,Wait(2)
;exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
;exten => 5678,3,Wait(2)
;exten => 5678,4,Playback(/tmp/asterisk-recording)
;exten => 5678,5,Wait(2)
;exten => 5678,6,Hangup

Once you make a recording, it needs to be moved to /var/lib/asterisk/sounds/custom with a new filename.gsm, e.g. mv /tmp/asterisk-recording.gsm /var/lib/asterisk/sounds/custom/hihoney.gsm. Then change the ownership of the file: chown asterisk:asterisk /var/lib/asterisk/sounds/custom/hihoney.gsm. You then can play the recording with a line like this in your dialplan: exten=>s,1,Playback(custom/hihoney) where hihoney is the name you assigned to the recording without its .gsm extension.

Where To Go From Here. After you get 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! 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.