We're just shy of the six month birthday for PBX in a Flash. So what better time to introduce version 1.2 which is chock full of new telephony goodies to whet your appetite for Internet Telephony. Tom King and Joe Roper have worked their usual Magic™ to come up with a pair of new ISOs that are nothing short of spectacular. Not only is PBX in a Flash leaner and meaner, but it's now incredibly flexible. You don't get the kitchen sink in PBX in a Flash ISOs. Instead you get a rock-solid CentOS 5.1 operating system on which to build an Internet telephony server that meets your specific needs. Want a 64-bit operating system? We've got it. Prefer to stick with a 32-bit operating system? We've got you covered there, too. Want to experiment with Asterisk® 1.6-beta? We've got it. Want to stick with Asterisk 1.4 for a production environment? We've got you covered. Do you prefer LVM, ext3, or SATA RAID for your disk drives? Well, take your pick. PBX in a Flash 1.2 now supports all of them. For those with a physical handicap, you now can install the complete system with no user intervention by typing ksauto at the first prompt. And, for PBX in a Flash development partners, we've even designed a 2-CD install set that makes generation of multiple systems with minimal Internet access a reality.
A Better Mousetrap. Asterisk-based LAMP aggregations, of course, are plentiful today, but we think we have a better mousetrap. Here are a few reasons why? First, PBX in a Flash is the only distribution that is totally source-based with Asterisk compiled from source. What that means is when you purchase add-on hardware and it has a problem for some reason, all of the tools are already in place for you to contact the manufacturer or reseller and have them reconfigure or recompile whatever is necessary on your system to get you back in business quickly. It also means that most of our applications are compiled from source on your specific hardware which assures a more reliable and stable software platform on which to build your telephony system.
Second, we don't release PBX in a Flash ISOs every other week. We don't have to. Every time a new security patch is released for Asterisk, the "other guys" have to create a new RPM or ISO to support it. That means your system is vulnerable while this process is underway. In many cases, it means reinstalling a new ISO and starting over. I wish I had a nickel for every time I reinstalled and basically started over with Asterisk@Home or trixbox. With PBX in a Flash, you simply type update-source at the command prompt and your system is brought current without missing a beat. The total downtime for your system is typically under 15 minutes!
Third, PBX in a Flash uses a two-step install process that all but eliminates the ISO obsolescence issues that have plagued other distributions. The PBX in a Flash ISO is used to install either the 32-bit or the 64-bit CentOS 5.1 operating system. When that process completes, the installer then searches multiple sites on the Internet for our "payload file" which contains the latest, greatest version of Asterisk which is compiled on-the-fly. The payload script also installs FreePBX and many of the customized features that make PBX in a Flash unique. If you need additional functionality, we have an entire web site, pbxinaflash.org, dedicated to add-on scripts, and it also has gotten a facelift. And, by the way, our typical add-on script installs without user intervention in under a minute. So... install what you need and skip the BloatWare. Using this design, most bugs are eliminated as well without your having to do much of anything. Translation: More siesta time. Less all-nighters!
Here's another reason that all of this matters. This is a true story that will give you a good handle on the flexibility that our design strategy brings to the table. We quietly introduced PBX in a Flash 1.2 to our loyal fan club last Wednesday. Within an hour after its release, the Asterisk Development Team announced a security patch and distributed new versions of Asterisk 1.4 and 1.6. Tom King, who was responsible for development of our latest payload files, happened to be scuba diving in the Atlantic Ocean as all of this unfolded. We sent him a text message to alert him to the problem. When Tom came up for some fresh air, he got the message. Then, using a cellphone from his boat, he kicked off an update script that regenerated all of the payload files with the latest Asterisk 1.4 and 1.6 security patches. And 90 minutes after the Asterisk security announcement, new PBX in a Flash 1.2 installs included both new versions of Asterisk. For those that installed their systems within the first 90 minutes, update-sources did the trick.
So today we're proud to introduce the 1.2 release of PBX in a Flash for Linux, Windows, and Macs. It's still the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users. Text-to-speech works, Bluetooth works, FreePBX 2.4 is rock-solid, the platform is open, and there already are custom install scripts for both Asterisk 1.4 and Asterisk 1.6 with many more just around the corner, perhaps as soon as this weekend.
As some of our regular readers know, we have been very concerned with the Asterisk development strategy that continues the process of regularly deleting commands and syntaxes with each major version change. Many of us rely upon these commands in building dialplans and vertical market applications for Asterisk so it causes a mess. PBX systems break that used to work. When that happens almost annually, it's a bad thing. One way that we hope to improve the dialogue with the developers is to make it easy for more people to experiment with Asterisk 1.6. Whether you choose our 32-bit or 64-bit ISO, you also have the option to install the latest Asterisk 1.6-beta and get involved in the process. Otherwise, we might as well look forward to annual train wrecks because the developers and Digium don't appear to be budging from their design strategy. You can read all about it here and here. We'll have more to say about it in coming weeks. For today, we're going to keep our sense of humor and walk you through the typical installation scenario to bring up a new PBX in a Flash 1.2 system with the latest version of Asterisk 1.4. When we're finished, you'll have a rock-solid telephony system to begin your Asterisk adventure. So let's get started.
Getting Started with PBX in a Flash 1.2. Begin by downloading either the 32-bit or 64-bit ISO image for PBX in a Flash. Don't worry. If you try to run the 64-bit install on a system that doesn't support it, it'll just sit there so you've got nothing to lose by trying the Ferrari first. As new locations for ISO downloads come on line, we will add them to the download list. Australia came on line yesterday thanks to Jim Lam. So just click on the location nearest to you, and you're off to the races. Once you've got the ISO image in hand, use your favorite tool to burn it to a bootable CD. This next step is the most important. Don't begin your installation until you first download and read Tom King's Installation Guide. It's an easy, non-technical read and will condense the install process to about 30 minutes. There also are loads of other helpful tutorials that are free for the downloading from our support site.
If you're new to all of this, let us recommend you try one of the $199 Everex Green PCs. Both WalMart and Egghead sell them, and they're just about perfect for a home or small business telephony server. And they're much less expensive to operate as well as being environmentally friendly. Just insert the CD containing the pbxinaflash.iso and then reboot the machine you wish to dedicate to PBX in a Flash. After reading Tom's tutorial and the initial prompts and warnings, choose an option and press the <Enter key> to begin the installation. If you want to first check the media for corruption, type linux mediacheck and then press the <Enter> key. When prompted, be sure to choose the option that erases all existing partitions and uses the default partition layout. Then choose your time zone and leave the UTC system clock option unchecked. Next choose a root password for your new system. Make it secure, and write it down. We plan to use this password for virtually everything on your new system. The install process begins. This includes MySQL, Apache, PHP, CUPS, Samba, WebMin, Subversion, SendMail, Yum, Bluetooth support, SSL, Perl, Python, the kernel development package, and much more. In about 15 minutes depending upon the speed of your PC, the install will pause to allow you to eject the CD. Click the Proceed button to continue after removing the CD. You must have an Internet connection now to complete the install so plug in a 10/100 cable if you haven't done so already. After reboot, the system will start up with CentOS 5.1, then download and install Asterisk and FreePBX, and search for the necessary installation script and payload file on pbxinaflash.net. If that site happens to be down, the script will go to pbxinaflash.com for the same payload file. Just to repeat, if you don't have Internet connectivity, then the installation cannot complete. When the installation finishes, reboot your system and log in as root. The IP address of your PBX in a Flash system will be displayed once you log in. If it's blank, type service network restart after assuring that you have Internet connectivity and access to a DHCP server that hands out IP addresses. Typing ifconfig should display your IP address on the eth0 port. Write it down. We'll need it in a minute.
Now that you've logged in as root, you should see the IP address displayed with the following command prompt: root@pbx:~/. If instead you see bash displayed as the command prompt and it's not green, then the installation has not completed successfully. This is probably due to network problems but also could be caused by the time being set incorrectly on your server. You can't compile Asterisk if the time on your computer is a date in the past! For this glitch you have to start over. If it's a network issue, fix it and then reboot and watch for the eth0 connection to complete. Assuming it doesn't fail the second time around, the installation will continue. Likewise, if you do not have DHCP on your network, the installation will fail because the PBX will not be given an IP address. Simply type netconfig, fill in the blanks and reboot. Tom's Guide goes into more troubleshooting detail. The install will recommence.
Required Steps to Complete the Install. There are four important things to do to complete the installation. First, from the command prompt, run genzaptelconf. This sets up your ZAP hardware as well as a timing source for conferencing. If you're using additional hardware for your Asterisk system, we recomend removing the 56K modem when you install the cards. This will help avoid interrupt conflicts. Second, decide how to handle the IP address for your PBX in a Flash server. The default is DHCP, but you don't want the IP address of your PBX changing. Phones and phone calls need to know how to find your PBX, and if your internal IP address changes because of DHCP, that's a problem. You have two choices. Either set your router to always hand out the same DHCP address to your PBX in a Flash server by specifying its MAC address in the reserved IP address table of your router, or run netconfig at the command prompt and assign a permanent IP address to your server. Be aware that netconfig no longer is a part of CentOS 5.1. We added it back in as part of the install. If you update your CentOS configuration, you will need to reinstall it by running update-scripts, then update-fixes, and then install-netconfig. If you experience problems with the process, see this message thread on the forum. The third configuration requirement probably accounts for more beginner problems with Asterisk systems than everything else combined. Read the next section carefully and do it now!
Getting Rid of One-Way Audio. There are some settings you'll need to add to /etc/asterisk/sip_custom.conf if you want to have reliable, two-way communications with Asterisk: nano -w /etc/asterisk/sip_custom.conf. The entries depend upon whether your Internet connection has a fixed IP address or a DHCP address issued by your provider. In the latter case, you also need to configure your router to support Dynamic DNS (DDNS) using a service such as dyndns.org. If you have a fixed IP address, then enter settings like the following using your actual public IP address and your private IP subnet:
localnet=192.168.1.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
If you have a public address that changes and you're using DDNS, then the settings would look something like the following:
localnet=192.168.0.0/255.255.255.0 (NOTE: The first 3 octets need to match your private IP addresses!)
Once you've made your entries, save the file: Ctrl-X, Y, then Enter. Reload Asterisk: amportal restart. If you assigned a permanent IP address, reboot your server: shutdown -r now.
Be aware that some people experience problems with the externhost approach outlined above. If your provider only gives you a dynamic IP address, you still can use the externip apprach above so long as you have a method to frequently verify your IP address. The approach we actually use on our home network is to run a little script every 5 minutes. If it finds that your outside IP address has changed, it will automatically update your sip_custom.conf file with the new address. To use our approach, create a file in /var/lib/asterisk/agi-bin names ip.sh. Here's the code:1
externip=`ping -c 1 $fqdn | cut -f 2 -d "(" | cut -f 1 -d ")" -s ↩
| grep -m 1 ^`
if [ -e /tmp/$externip ] ; then
echo No IP Update Required ;
echo IP Update Required ;
touch /tmp/$externip ;
echo "externip=$externip" > /etc/asterisk/sip_custom.conf
echo "localnet=$localnet/255.255.255.0" >> /etc/asterisk/sip_custom.conf
asterisk -rx "dialplan reload" ;
On line 2 of the above code, enter the fully-qualified domain name for your server that is registered with your DDNS host. Take a look at this thread for information on DNS-O-Matic which is free.
On line 3, enter the internal subnet for your server. This is usually 192.168.0.0 or 192.168.1.0. YMMV!
Save the file and give it execute permissions: chmod +x /var/lib/asterisk/agi-bin/ip.sh. Then make asterisk the file owner: chown asterisk:asterisk /var/lib/asterisk/agi-bin/ip.sh.
Finally, add the following entry to the bottom of /etc/crontab:
*/5 * * * * asterisk /var/lib/asterisk/agi-bin/ip.sh > /dev/null
Getting Your Machine Up to Date. Tom King, one of our lead developers, has gone to great pains to make it easy for you to always have a current system. All you have to do is type a few commands, but you do have to type them. So do it now! After logging in as root, type update-scripts to get the latest PBX in a Flash scripts installed on your system. This doesn't run them, it merely makes them available for you to run them. Once you complete this step, you can always review the latest scripting options by typing help-pbx. Now run update-fixes to apply the latest patches to your PBX in a Flash system. When it completes, you're up to date. If you want the latest version of Asterisk, it's easy! Just run update-source. In the case of PBX in a Flash 1.2, you have the latest version of Asterisk 1.4 or 1.6-beta... at least for today.
Activating Email Delivery of Voicemail Messages. We've previously shown how to configure systems to reliably deliver email messages whenever a voicemail arrives unless your ISP happens to block downstream SMTP mail servers. Here's the link in case you need it. As it happens, you really don't have to use a real fully-qualified domain name to get this working. So long as the entry (such as pbx.dyndns.org) is inserted in both the /etc/hosts file and /etc/asterisk/vm_general.inc with a matching servermail entry of email@example.com (as explained in the link above), your system will reliably send emails to you whenever you get a voicemail if you configure your extensions in freePBX to support this capability. You can, of course, put in real host entries if you prefer. For 90% of the systems around the world, if you just want your server to reliably e-mail you your voicemail messages, make line 3 of /etc/hosts look like this with a tab after 127.0.0.1 and spaces between the domain names:
127.0.0.1 pbx.dyndns.org pbx.local pbx localhost.localdomain localhost
And then make line 6 of /etc/asterisk/vm_general.inc look like the following:
Now issue the following two commands to make the changes take effect:
service network restart
The command "setup-mail" can be used from the Linux prompt to set the fully-qualified domain name (FQDN) of the mail that is sent out from your server. This may help mail to be delivered from the PBX. One of things mail servers do to reduce spam is to do a reverse lookup on where the mail has come from, checking that there is actually a mailserver at the other end. You can only do this if you have set up dynamic DNS or if you have pointed a hostname at your fixed IP address. Once you have done this, and assuming your ISP is cooperative, then you will receive your voicemails via email if you wish (this is set within FreePBX),and your PBX will email you when FreePBX needs an update. You set this feature in FreePBX General Settings.
If your hosting provider blocks downstream SMTP servers to reduce spam, here's a simple way to use your gMail account (free!) as your SMTP Relay Host. Then you never have to worry about this again!
Setting Passwords and Other Stuff. Be aware that there is a major security hole in FreePBX. Using FreePBX admin security alone will not protect your system from a web attack and may compromise root access to your entire server. For this reason, we recommend that you log in as root and immediately run passwd-master. This establishes Apache htaccess security on your FreePBX web interface. After running this conversion utility, you can only log into the FreePBX admin interface with the username maint (not admin) and the password which you establish when you run the utility.
Other passwords can be set in your system with these commands:
passwd... reset your root user password
passwd-maint... reset your FreePBX maint password
passwd-wwwadmin... for users needing FOP and MeetMe access
passwd-meetme... for users needing only MeetMe access
passwd-webmin... for users needing WebMin access to your server (very dangerous!)
There's also an Administration password that you can set in the KennonSoft UI that displays when you point your browser to the IP address of your server. Do NOT use the same password here that you use elsewhere as it is not overly secure.
Configuring WebMin. WebMin is the Swiss Army Knife of Linux. It provides TOTAL access to your system through a web interface. Search Nerd Vittles for webmin if you want more information. Be very careful if you decide to enable it on the public Internet. You do this by opening port 9001 on your router and pointing it to the private IP address of your PBX in a Flash server. Before using WebMin, you need to set up a username and password for access. From the Linux prompt while logged in as root, type the following command where admin is the username you wish to set up and foo is the password you've chosen for the admininstrator account. HINT: Don't use admin and foo as your username and password for WebMin unless you want your server trashed!
/usr/libexec/webmin/changepass.pl /etc/webmin root password
To access WebMin on your private network, go to http://192.168.0.123:9001 where 192.168.0.123 is the private IP address of your PBX in a Flash server. Then type the username and password you assigned above to gain entry. To stop WebMin: /etc/webmin/stop. To start WebMin: /etc/webmin/start. For complete documentation, go here.
Updating and Configuring FreePBX. FreePBX 2.4 is installed as part of the PBX in a Flash 1.2 implementation. This incredible, web-based tool provides a complete menu-driven user interface to Asterisk. The entire FreePBX project is a model of how open source development projects ought to work. And having Philippe Lindheimer's as the Captain of the Ship is just icing on the cake. All it takes to get started with FreePBX is a few minutes of configuration, and you'll have a functioning Asterisk PBX complete with voicemail, music on hold, call forwarding, and a powerful interactive voice response (IVR) system. There is excellent documentation for FreePBX which you should read at your earliest convenience. It will answer 99% of your questions about how to use and configure FreePBX. For the one percent that is not covered in the Guide, visit the FreePBX Forums which are frequented regularly by the FreePBX developers. Kindly post FreePBX questions on their forum rather than the PBX-in-a-Flash Forum. This helps everybody. Now let's get started.
NOTE: PBX in a Flash comes with the IPtables firewall enabled on your system. If this causes problems with access to the FreePBX repository (for loading the FreePBX updates below), you can easily (and temporarily) turn off the firewall. Type help-pbx for assistance. Don't forget to restart the firewall especially if your system has any Internet exposure!
Now move to a PC or Mac and, using your favorite web browser, go to the IP address you deciphered above for your new server. Be aware that FreePBX has a difficult time displaying properly with IE6 and IE7 and regularly blows up with older versions of Safari. Be safe. Use Firefox. From the PBX in a Flash Main Menu in your web browser, click on the Administration link and then click the FreePBX button. The username and password both default to admin. Click Apply Configuration Changes, Continue with Reload, and then Refresh your browser screen. Now click the Module Administration option in the left frame once FreePBX loads. Now click Check for Updates online in the upper right panel. Next, click Download All which will select every module for download and install. The important step here is to move down the list and Deselect Speed Dials and PHPAGI from the download and install options. Once these apps have been deselected, scroll to the bottom of the page and click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. Now repeat the process once more and do not deselect the two applications, then Process, Confirm, Return, Apply Config Changes, and Continue with Reload. Finally, scroll down the Modules listing until you get to the Maintenance section. Click on each of the following and choose Install: ConfigEdit, Sys Info, and phpMyAdmin. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. All three of these tools now are installed in the Maintenance section of the Tools tab of FreePBX. One final step, and you're good to go. An update of FreePBX has been released. Click Check for Updates online. Then choose Download and Upgrade for the Core, FreePBX Framework, and System Dashboard modules. Then click Process, then Confirm, then Return once the apps are downloaded and installed, then Apply, then Continue with Reload. You now have an up-to-date version of FreePBX. You'll need to repeat the drill every few weeks as new updates are released. This will assure that you have all of the latest and greatest software. To change your Admin password, click on the Setup tab in the left frame, then click Administrators, then Admin in the far right column, enter a new password, and click Submit Changes, Apply Configuration Changes, and Continue with reload. We're going to be repeating this process a number of times in the next section so... when instructed to Save Your Changes, that means "click Submit Changes, Apply Configuration Changes, and Continue with reload."
Choosing Internet Telephony Hosting Providers for Your System. Before you can place calls to users outside your system or to receive incoming calls, you'll need at least one provider (each) for your incoming phone number (DID) and incoming calls as well as a provider for your outbound calls (terminations). We have a list of some of our favorites here, and there are many, many others. You basically have two choices with most providers. You can either pay as you go or sign up for an all-you-can-eat plan. Most of the latter plans also have caps on minutes so it's more akin to all-they-care-for-you-to-eat, and there are none of the latter plans for business service. In the U.S. market, the going rate for pay as you go service is about 1.5¢ per minute rounded to the tenth of a minute. The best deal on DIDs is from les.net. They charge $3.99 a month for a DID with unlimited, free incoming calls. WARNING: Before you sign up for any all-you-can-eat plan, do some reading about the service providers. Some of them are real scam artists with backbilling and all sorts of unconscionable restrictions. You need to be careful. Our cardinal rule in the VoIP Wild West is never, ever entrust your entire PBX to a single hosting provider. As Forrest Gump would say, "Stuff happens!" And life's too short to have dead telephones, even if it's a rarity.
Setting Up FreePBX to Make Your First Call. There are four components in FreePBX that need to be configured before you can place a call or receive one from outside your PBX in a Flash system. So here's FreePBX for Dummies in less than 50 words. You need to configure Trunks, Extensions, Outbound Routes, and Inbound Routes. Trunks are hosting provider specifications that get calls delivered to and transported from your PBX to the rest of the world. Extensions are internal numbers on your PBX that connect your PBX to telephone hardware or softphones. Inbound Routes specify what should be done with calls coming in on a Trunk. Outbound Routes specify what should be done with calls going out to a Trunk. Everything else is bells and whistles.
Trunks. When you sign up with most of the better ITHP's that support Asterisk, they will provide documentation on how to connect their service with your Asterisk system. If they have a trixbox tutorial, use that since it also uses FreePBX as the web front end to Asterisk. Here's an example from les.net. And here's the Vitelity support page although you will need to set up an account before you can access it. We also have covered the setups for a number of providers in previous articles. Just search the Nerd Vittles site for the name of the provider you wish to use. You'll also find many Trunk setups in the trixbox Trunk Forum. Once you find the setup for your provider, add it in FreePBX by going to Setup, Trunks, Add SIP Trunk. Our AxVoice setup (which is all entered in the Outgoing section with a label of axvoice) looks like this with a Registration String of yourusername:firstname.lastname@example.org:
And our Vitelity Outbound Trunk looks like the following (labeled vitel-outbound) with no registration string:
Extensions. Now let's set up a couple of Extensions to get you started. A good rule of thumb for systems with less than 50 extensions is to reserve the IP addresses from 192.x.x.201 to 192.x.x.250 for your phones. Then you can create extension numbers in FreePBX to match those IP addresses. This makes it easy to identify which phone on your system goes with which IP address and makes it easy for end-users to access the phone's GUI to add bells and whistles. To create extension 201 (don't start with 200), click Setup, Extensions, Generic SIP Device, Submit. Then fill in the following blanks leaving the defaults in the other fields for the time being.
User Extension ... 201
Display Name ... Home
Outbound CID ... [your 10-digit phone number if you have one; otherwise, leave blank]
Emergency CID ... [your 10-digit phone number for 911 ID if you have one; otherwise, leave blank]
secret ... 1234
dtmfmode ... rfc2833
Voicemail & Directory ... Enabled
voicemail password ... 1234
email address ... email@example.com [if you want voicemail messages emailed to you]
pager email address ... firstname.lastname@example.org [if you want to be paged when voicemail messages arrive]
email attachment ... yes [if you want the voicemail message included in the email message]
play CID ... yes [if you want the CallerID played when you retrieve a message]
play envelope ... yes [if you want the date/time of the message played before the message is read to you]
delete Vmail ... yes [if you want the voicemail message deleted after it's emailed to you]
vm options ... callback=from-internal [to enable automatic callbacks by pressing 3,2 after playing a voicemail message]
vm context ... default
Now create several more extensions using the template above: 202, 203, 204, and 205 would be a good start. Keep the passwords simple. You'll need them whenever you configure your phone instruments.
Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. We're going to skip that tutorial today. You can search the site for lots of information on choosing providers. Assuming you have only one or two for starters, let's just set up a default outbound route for all your calls. Using your web browser, access FreePBX on your server and click Setup, Outbound Routes. Enter a route name of Everything. Enter the dial patterns for your outbound calls. In the U.S., you'd enter something like the following:
Click on the Trunk Sequence pull-down and choose your providers in the order you'd like them to be used for outbound calls.Click Submit Changes and then save your changes. Note that a second choice in trunk sequence only gets used if the calls fail to go through using your first choice. You'll notice there's already a 9_outside route which we don't need. Click on it and then choose Delete Route 9_outside. Save your changes.
Inbound Routes. We're also going to abbreviate the inbound routes tutorial just to get you going quickly today. The idea here is that you can have multiple DIDs (phone numbers) that get routed to different extensions or ring groups or departments. For today, we recommend you first build a Ring Group with all of the extension numbers you have created. Once you've done that, choose Inbound Routes, leave all of the settings at their default values and move to the Set Destination section and choose your Ring Group as the destination. Now click Submit and save your changes. That will set up a default incoming route for your calls. As you add bells and whistles to your system, you can move the Default Route down the list of priorities so that it only catches calls that aren't processed with other inbound routing rules.
General Settings. Last, but not least, we need to enter an email address for you so that you are notified when new FreePBX updates are released. Scroll to the bottom of the General Settings screen after selecting it from the left panel. Plug in your email address, click Submit, and save your changes. Done!
Adding Plain Old Phones. Before your new PBX will be of much use, you're going to need something to make and receive calls, i.e. a telephone. For today, you've got several choices: a POTS phone, a softphone, or a SIP phone. Option #1 and the best home solution is to use a Plain Old Telephone or your favorite cordless phone set (with 8-10 extensions) if you purchase a little device known as a Sipura SPA-3102. It's under $70. Be sure you specify that you want an unlocked device, meaning it doesn't force you to use a particular service provider. This device also supports connection of your PBX to a standard office or home phone line as well as a telephone.
Downloading a Free Softphone. Unless you already have an IP phone, the easiest way to get started and make sure everything is working is to install an IP softphone. You can download a softphone for Windows, Mac, or Linux from CounterPath. Or download the pulver.Communicator or the snom 360 Softphone which is a replica of perhaps the best IP phone on the planet. Here's another great SIP/IAX softphone for all platforms that's great, too, and it requires no installation: Zoiper 2.0 (formerly IDEfisk). All are free! Just install and then configure with the IP address of your PBX in a Flash server. For username and password, use one of the extension numbers and passwords which you set up with freePBX. Once you make a few test calls, don't waste any more time. Buy a decent SIP telephone. We think the best value in the marketplace with excellent build quality and feature set (but probably not the best sound quality) is the $79 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. Our personal favorite and the phone that PBX in a Flash officially supports is the Aastra 57i or 57iCT which also includes cordless DECT phone. Do some reading before you buy. The Voxilla forums are a good place to start.
A Word About Ports. For the techies out there that want "the rest of the story" to properly configure firewalls, here's a list of the ports available and used by PBX in a Flash:
TCP 80 - HTTP
TCP 9080 - Duplicate HTTP
TCP 22 - SSH
TCP 9022 - Duplicate SSH
TCP 9001 - WebMin
UDP 10000-20000 - RTP
UDP 5004-5082 - SIP
UDP 4569 - IAX2
UDP 2727 - Media Gateway
Where To Go From Here. The PBX in a Flash script repository at pbxinaflash.org also has gotten a facelift. That should be your next stop because it is the home of all the goodies that make PBX in a Flash shine. Tom King, the ultimate scripting guru, manages that site. So check it often. And now that PBX in a Flash 1.2 is out the door, we've been chomping at the bit to get all of our Nerd Vittles Goodies ported over. Most of our original collection work flawlessly with Asterisk 1.4 including AsteriDex, Yahoo News Headlines, Weather by Airport Code, Weather by Zip Code, Worldwide Weather Forecasts, Telephone Reminders, MailCall for Asterisk, and TeleYapper. We have not yet completed testing with Asterisk 1.6... which has a text-to-speech impairment at the moment. Complete documentation for each application also is provided at the link above. And, if you still have a DBT-120 Bluetooth adapter, you'll be happy to learn that it works out-of-the-box with PBX in a Flash on your new Everex Green PC. Dust off our recent article on Proximity Detection, and you should be in business in under 10 minutes. Enjoy!
Awesome Vitelity Special. Vitelity has generously offered a terrific discount for Nerd Vittles readers. You now can get an almost half-price DID from our special Vitelity sign-up link. If you're seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. When you use our special link to sign up, Nerd Vittles gets a few shekels down the road to support our open source development efforts while you get an incredible signup deal as well. The going rate for Vitelity's DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For our users, here's a deal you can't (and shouldn't) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls and four simultaneous channels for just $3.99 a month. To check availability of local numbers and tiers of service from Vitelity, click here. NOTE: You can only use the Nerd Vittles sign-up link to order your DIDs, or you won't get the special pricing! Vitelity's rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage. Any balance is refundable if you decide to discontinue service with Vitelity.
Some Recent Nerd Vittles Articles of Interest...
- Join the following line to the original line of code whenever you encounter the ↩ character. [↩]