PIAF 101: Taking Asterisk 10 for a Spin
There’s been some interest in a quick-and-dirty guide to get PBX in a Flash up and running without much in the way of bells and whistles. So here you go. This step-by-step will get PIAF-Red with Asterisk® 10 or PIAF-Purple with Asterisk 1.8.5.0 or 1.8.6.01 humming away. If you’re going to do things this way, then make sure your PIAF server or virtual host with PIAF is sitting behind a secure, hardware-based firewall (such as dLink’s Gaming Router) with NO INTERNET PORT EXPOSURE to your PIAF box!
UPDATE: Digium has dropped support for Google Voice in Asterisk 10 so we no longer recommend Asterisk 10 for production use. You can read all about it here.
Atom-based PC Platform. For the least expensive hardware alternative, pick up an Atom-based PC, preferably not an EEE PC because of the network driver incompatibility with CentOS. The refurbished Revos work fine. Someone has actually tested them! And they can easily support a small business with dozens of phones.
PIAF ISO Setup. Once you have your hardware connected to a reliable Internet source, you’ll need to choose the appropriate ISO for your hardware. If you have a CD-ROM or DVD drive on your server, we’d recommend the 32-bit PIAF 1.7.5.6.3 ISO. Just download it from SourceForge or one of the PIAF mirror sites, burn it to a CD, and then boot your server from the CD. If your server lacks a CD-ROM and DVD drive, then download the brand new 32-bit PIAF 1.7.5.6.3 Flash-Only ISO from SourceForge and copy it to a 1GB or larger thumb drive following the instructions in this Nerd Vittles tutorial. Then boot your server from the thumb drive. You’ll find OpenVZ and VMware templates on our download mirrors as well.
PIAF Installation. Once you’ve booted the PIAF installer, you’ll be prompted to choose an installation method. For most users, simply pressing the Enter key will get things started. Choose a keyboard and time zone when prompted and then enter a very secure root password for your new server. The installer then will load CentOS 5.6 onto your server. When complete, your server will reboot. Remove the CD or Flash Drive at this point, and you’ll be prompted to choose the version of Asterisk to install. Just for fun, choose PIAF-Red which loads the latest Asterisk 10 beta. It works just fine!
During the final phase of the install, you will be prompted to choose a master password for FreePBX® and the other VoIP web utilities. Once your server reboots, log into the Linux CLI using your root password and write down the IP address of your server from the status display.
FreePBX Setup. Most of your life with PBX in a Flash will be spent using the FreePBX web GUI and your favorite browser. Just click on the image below to enlarge. To access the FreePBX GUI, point your browser at the IP address you wrote down. Read the RSS Feed in the PIAF GUI for late-breaking security alerts. Then click on the Users button which will toggle to the Admin menu. Click the FreePBX icon. When prompted for your username and password, the username is maint. The password will be the FreePBX master password you chose in completing the PIAF install.
To get a minimal system functioning, here’s the 5-minute drill. You’ll need to set up at least one extension with voicemail, configure a free Google Voice account for free calls in the U.S. and Canada, configure inbound and outbound routes to manage incoming and outgoing calls, and plug your maint password into CallerID Superfecta so that names arrive with your incoming calls. Once you add a phone with your extension credentials, you’re done.
Extension Setup. Now let’s set up an extension 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 USING VERY SECURE PASSWORDS and 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]Device Options
secret … 1299864Xyz [make this unique AND secure!]
dtmfmode … rfc2833
Voicemail & Directory … Enabled
voicemail password … 14332 [make this unique AND secure!]
email address … yourname@yourdomain.com [if you want voicemail messages emailed to you]
pager email address … yourname@yourdomain.com [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
Write down the passwords. You’ll need them to configure your SIP phone.
Extension Security. We cannot overstress the need to make your extension passwords secure. All the firewalls in the world won’t protect you from malicious phone calls on your nickel if you use your extension number or something like 1234 for your extension password if your SIP or IAX ports happen to be exposed to the Internet. Incredible PBX automatically randomizes all of the extension passwords for you.
In addition to making up secure passwords, the latest versions of FreePBX also let you define the IP address or subnet that can access each of your extensions. Use it!!! Once the extensions are created, edit each one and modify the permit field to specify the actual IP address or subnet of each phone on your system. A specific IP address entry should look like this: 192.168.1.142/255.255.255.255. If most of your phones are on a private LAN, you may prefer to use a subnet entry in the permit field like this: 192.168.1.0/255.255.255.0 using your actual subnet.
Adding a Google Voice Trunk. There are lots of trunk providers, and one of the real beauties of having your own PBX is that you don’t have to put all of your eggs in the same basket… unlike the AT&T days. We would encourage you to take advantage of this flexibility. With most providers, you don’t pay anything except when you actually use their service so you have nothing to lose.
For today, we’re going to take advantage of Google’s current offer of free calling in the U.S. and Canada through the end of this year. You also get a free phone number in your choice of area codes. PBX in a Flash now installs a Google Voice module for FreePBX that lets you set up your Google Voice account with PBX in a Flash in just a few seconds once you have your credentials.
Signing Up for Google Voice. You’ll need a dedicated Google Voice account to support PBX in a Flash. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So keep this account a secret!
We’ve tested this extensively using an existing Gmail account rather than creating a separate account. Take our word for it. Inbound calling is just not reliable. The reason seems to be that Google always chooses Gmail chat as the inbound call destination if there are multiple registrations from the same IP address. So… set up a dedicated Gmail and Google Voice account, and use it exclusively with PBX in a Flash. Google Voice no longer is by invitation only. If you’re in the U.S. or have a friend that is, head over to the Google Voice site and register. If you’re living on another continent, see MisterQ’s posting for some tips on getting set up.
You must choose a telephone number (aka DID) for your new account, or Google Voice calling will not work… in either direction. You also have to tie your Google Voice account to at least one working phone number as part of the initial setup process. Your cellphone number will work just fine. Don’t skip this step either. Just enter the provided confirmation code when you tell Google to place the test call to the phone number you entered. Once the number is registered, you can disable it if you’d like in Settings, Voice Setting, Phones. But…
IMPORTANT: Be sure to enable the Google Chat option as one of your phone destinations in Settings, Voice Setting, Phones. That’s the destination we need for PBX in a Flash to function with Google Voice! Otherwise, inbound and/or outbound calls will fail. If you don’t see this option, you may need to call up Gmail and enable Google Chat there first. Then go back to the Google Voice Settings and enable it. Be sure to try one call each way from Google Chat in Gmail. Then disable Google Chat in GMail for this account. Otherwise, it won’t work with PIAF.
While you’re still in Google Voice Settings, click on the Calls tab. Make sure your settings match these:
- Call Screening – OFF
- Call Presentation – OFF
- Caller ID (In) – Display Caller’s Number
- Caller ID (Out) – Don’t Change Anything
- Do Not Disturb – OFF
- Call Options (Enable Recording) – OFF
- Global Spam Filtering – ON
Click Save Changes once you adjust your settings. Under the Voicemail tab, plug in your email address so you get notified of new voicemails. Down the road, receipt of a Google Voice voicemail will be a big hint that something has come unglued on your PBX.
Configuring Google Voice Trunk in FreePBX. All trunk configurations now are managed within FreePBX, including Google Voice. This makes it easy to customize PBX in a Flash to meet your specific needs. Click the Setup tab and choose Google Voice in the Third Party Addons. To Add a new Google Voice account, just fill out the form:
Phone number is your 10-digit Google Voice number. Username is your Google Voice account name without @gmail.com. NOTE: You must use a Gmail.com address in the current version of this module! Password is your Google Voice password. NOTE: Don’t use 2-stage password protection in this Google Voice account! Be sure to check all three boxes: Add trunk, Add routes, and Agree to TOS. Then click Submit Changes and reload FreePBX. Down the road, you can add additional Google Voice numbers by clicking Add GoogleVoice Account option in the right margin and repeating the drill. For Google Apps support, see this post on the PIAF Forum.
Outbound Routes. The idea behind multiple outbound routes is to save money. Some providers are cheaper to some places than others. It also provides redundancy which costs you nothing if you don’t use the backup providers. The Google Voice module actually configures an Outbound Route for 10-digit Google Voice calling as part of the automatic setup. If this meets your requirements, then you can skip this step for today.
Inbound Routes. An Inbound Route tells PBX in a Flash how to route incoming calls. 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’ll build a simple route that directs your Google Voice calls to extension 201. Choose Inbound Routes, leave all of the settings at their default values except enter your 10-digit Google Voice number in the DID Number field. Enable CallerID lookups by choosing CallerID Superfecta in the CID Lookup Source pulldown. Then move to the Set Destination section and choose Extensions in the left pull-down and 201 in the extension pull-down. Now click Submit and save your changes. That will assure that incoming Google Voice calls are routed to extension 201.
IMPORTANT: Before Google Voice calling will actually work, you must restart Asterisk from the Linux command line interface. Log into your server as root and issue this command: amportal restart.
CallerID Superfecta Setup. CallerID Superfecta needs to know your maint password in order to access the necessary modules to retrieve CallerID information for inbound calls. Just click Setup, CID Superfecta, and click on Default in the Scheme listings in the right column. Scroll down to the General Options section and insert your maint password in the Password field. You may also want to enable some of the other providers and adjust the order of the lookups to meet your local needs. Click Agree and Save once you have the settings adjusted.
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.
Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone such as the $50 Nortel color videophone we’ve recommended previously. You’ll also find lots of additional recommendations on Nerd Vittles and in the PBX in a Flash Forum. If you’re like us, we want to make damn sure this stuff works before you shell out any money. So, for today, let’s download a terrific (free) softphone to get you started. We recommend X-Lite because there are versions for Windows, Mac, and Linux. So download your favorite from this link. Install and run X-Lite on your Desktop. At the top of the phone, click on the Down Arrow and choose SIP Account Settings, Add. Enter the following information using 201 for your extension and your actual password for extension 201. Then plug in the actual IP address of your PBX in a Flash server instead of 192.168.0.251. Click OK when finished. Your softphone should now show: Available.
Enabling Google Voicemail. Some have requested a way to retain Google’s voicemail system for unanswered calls in lieu of using Asterisk voicemail. The advantage is that Google offers a free transcription service for voicemail messages. To activate this, you’ll need to edit the [googlein] context in extensions_custom.conf in /etc/asterisk. Just modify the last four lines in the context so that they look like this and then restart Asterisk: amportal restart
;exten => s,n(regcall),Answer
;exten => s,n,SendDTMF(1)
exten => s,n(regcall),Set(DIAL_OPTIONS=${DIAL_OPTIONS}aD(:1))
exten => s,n,Goto(from-trunk,gv-incoming,1)
But I Don’t Want to Use Google Voice. If you’d prefer not to use Google Voice at all with PBX in a Flash, that’s okay, too. Here’s how to disable it and avoid the chatter in the Asterisk CLI. Log into your server as root and edit /etc/asterisk/modules.conf. Change the first three lines in the [modules] context so that they look like this. Then restart Asterisk: amportal restart.
autoload=yes
noload => res_jabber.so
noload => chan_gtalk.so
Where To Go From Here. We’ve barely scratched the surface of what you can do with your new PBX in a Flash system. If you’re new to all of this, then your next step probably should be last week’s Incredible PBX 2.0 tutorial. It’s a 5-minute addition that installs nearly 50 Asterisk applications that will keep you entertained for the rest of the year. If you’d prefer to do it yourself, then… enjoy!
Originally published: Monday, August 29, 2011
Need help with Asterisk? Visit the PBX in a Flash Forum.
Or Try the New, Free PBX in a Flash Conference Bridge.
whos.amung.us If you’re wondering what your fellow man is reading on Nerd Vittles these days, wonder no more. Visit our new whos.amung.us statistical web site and check out what’s happening. It’s a terrific resource both for us and for you.
New Vitelity Special. Vitelity has generously offered a new discount for PBX in a Flash users. You now can get an almost half-price DID and 60 free minutes from our special Vitelity sign-up link. If you’re seeking the best flexibility in choosing an area code and phone number plus the lowest entry level pricing plus high quality calls, then Vitelity is the hands-down winner. Vitelity provides Tier A DID inbound service in over 3,000 rate centers throughout the US and Canada. And, when you use our special link to sign up, the Nerd Vittles and PBX in a Flash projects get a few shekels down the road while you get an incredible signup deal as well. The going rate for Vitelity’s DID service is $7.95 a month which includes up to 4,000 incoming minutes on two simultaneous channels with terminations priced at 1.45¢ per minute. Not any more! For PBX in a Flash users, here’s a deal you can’t (and shouldn’t) refuse! Sign up now, and you can purchase a Tier A DID with unlimited incoming calls for just $3.99 a month and you get a free hour of outbound calling to test out their call quality. To check availability of local numbers and tiers of service from Vitelity, click here. Do not use this link to order your DIDs, or you won’t get the special pricing! After the free hour of outbound calling, Vitelity’s rate is just 1.44¢ per minute for outbound calls in the U.S. There is a $35 prepay when you sign up. This covers future usage and any balance is fully refundable if you decide to discontinue service with Vitelity.Some Recent Nerd Vittles Articles of Interest…
- See this Nerd Vittles article for Asterisk 1.8.6.0 install instructions. [↩]
The Incredible PBX Inventory. For those that have never heard of The Incredible PBX, here’s a feature list of components you get in addition to the base install of PBX in a Flash the latest CentOS 5.x, Asterisk 1.8, FreePBX 2.8, and Apache, SendMail, MySQL, PHP, phpMyAdmin, IPtables Linux firewall, Fail2Ban, and WebMin. Cepstral TTS, Fax, Hamachi VPN, and Mondo Backups are just one command away and may be installed using some of the PBX in a Flash-provided scripts.
Google Voice Setup. You’ll need a dedicated Google Voice account to support The Incredible PBX. The more obscure the username (with some embedded numbers), the better off you will be. This will keep folks from bombarding you with unsolicited Gtalk chat messages, and who knows what nefarious scheme will be discovered using Google messaging six months from now. So why take the chance. Keep this account a secret!
RentPBX Setup. Once you have your Google Voice credentials, you’re ready to get your virtual machine at RentPBX set up. First, you’ll need an account. So visit RentPBX.com and sign up for an account using the coupon code above to get your discount. Pick a cloud server to host your new system, choose the PIAF-Purple 1.7.5.6 install option, set up a username and very secure password, and you’re done. Once your account is established and you receive your credentials, here’s the 5-minute procedure to install the special RentPBX-edition of Incredible PBX to begin making free calls in the U.S. and Canada through Google Voice.
The Gmail Notification Address is the email address where you wish to receive alerts when incoming and outgoing Google Voice calls are placed using The Incredible PBX. And your FreePBX maint Password is the very secure password you want to use to access FreePBX using a web browser. We need this password to properly configure the CallerID Superfecta for you. By the way, none of this confidential information ever leaves your machine… just in case you were wondering.
You’ll find a link at the top of the page. When the installer finishes, READ THE SCREEN just for grins.
Securing Your RentPBX Server. The WhiteList application is not yet supported in the cloud. So you’ll need to secure your system to avoid endless hack attempts on your SIP resources. Here’s how. First, write down the IP addresses of your RentPBX server and your home network. Second, print out your existing IPtables configuration. The file to print is /etc/sysconfig/iptables. Third, make a backup copy of the file. While logged into your server with SSH, the easiest way is like this:
Configuring a SIP Phone. There are hundreds of terrific SIP telephones and softphones for Asterisk-based systems. Once you get things humming along, you’ll want a real SIP telephone such as the 
Installing PBX in a Flash. Here's a quick tutorial to get PBX in a Flash installed. To use Incredible PBX for Asterisk 1.8, just install the latest 32-bit version of PBX in a Flash. Unlike other Asterisk aggregations, PBX in a Flash utilizes a two-step install process. The ISO only installs the CentOS 5.6 operating system. Once CentOS is installed, the server reboots and downloads a payload file that includes Asterisk, FreePBX, and many other VoIP and Linux utilities including all of the new Google Voice components. Just choose the new Purple Payload to get the latest Asterisk 1.8 release and all of the Google Voice goodies! 
Incredible PBX Test Flight. The proof is in the pudding as they say. So let's try two simple tests. First, let's place an outbound call. Using the softphone, dial your 10-digit cellphone number. Google Voice should transparently connect you. Answer the call and make sure you can send and receive voice on both phones. Second, from another phone, call the Google Voice number that you've dedicated to The Incredible PBX. Your softphone should begin ringing shortly. Answer the call, press 1 to accept the call, and then make sure you can send and receive voice on both phones. Hang up. If everything is working, congratulations! 
Solving One-Way Audio Problems. If you experience one-way audio on some of your phone calls, you may need to adjust the settings in /etc/asterisk/sip_custom.conf. Just uncomment the first two lines by removing the semicolons. Then replace 173.15.238.123 with your public IP address, and replace 192.168.0.0 with the subnet address of your private network. There are similar settings in gtalk.conf that can be activated although we've never had to use them. In fact, we've never had to use any of these settings. After making these changes, save the file(s) and restart Asterisk with the command: amportal restart.
Choosing a VoIP Provider for Redundancy. Nothing beats free when it comes to long distance calls. But nothing lasts forever. And, in the VoIP World, redundancy is dirt cheap. So we strongly recommend you set up another account with Vitelity using our special link below. This gives your PBX a secondary way to communicate with every telephone in the world, and it also gets you a second real phone number for your new system... so that people can call you. Here's how it works. You pay Vitelity a deposit for phone service. They then will bill you $3.99 a month for your new phone number. This $3.99 also covers the cost of unlimited inbound calls (two at a time) delivered to your PBX for the month. For outbound calls, you pay by the minute and the cost is determined by where you're calling. If you're in the U.S., outbound calls to anywhere in the U.S. are a little over a penny a minute. If you change your mind about Vitelity and want a refund of the balance in your account, all you have to do is ask. The trunks for Vitelity already are preconfigured with The Incredible PBX. Just insert your credentials using FreePBX. Then add the Vitelity trunk as the third destination for your default outbound route. That's it. Congratulations! You now have a totally redundant phone system.
Special Thanks. It's hard to know where to start in expressing our gratitude for all of the participants that made today's incredibly simple-to-use product possible. Please bear with us. To Mark Spencer, Malcolm Davenport, and the rest of the Asterisk development team, thanks for a much improved Asterisk. To Philippe Sultan and his co-developers, thank you for finally making Jabber jabber with Asterisk. To Leif Madsen, our special thanks for your early pioneering work with Gtalk and Jabber which got this ball rolling. To Philippe Lindheimer & Co., thanks for FreePBX 2.8 which really makes Asterisk shine. To Lefteris Zafiris, thank you for making Flite work with Asterisk 1.8 thereby preserving all of the Nerd Vittles text-to-speech applications. To Darren Sessions, thanks for whipping app_swift into shape and restoring Cepstral and commercial TTS applications to the land of the living with Asterisk 1.8. And to our pal, Tom King, we couldn't have done it without you. You rolled up your sleeves and really made CentOS 5.6 and Asterisk 1.8 sit up and bark. No one will quite understand what an endeavor that was until they try it themselves. You've made it look so easy. And, finally, to our dozens of beta testers, THANK YOU! We've implemented almost all of your suggestions.
Additional Goodies. Be sure to log into your server as root and look through the scripts added in the /root and /root/nv folders. You'll find all sorts of goodies to keep you busy. There's an all-new incrediblefax.sh script that painlessly installs and configures HylaFax and AvantFax for state-of-the-art faxing. The 32-bit install-cepstral script does just what it says. With Allison's Cepstral voice, you'll have the best TTS implementation for Asterisk available. ipscan is a little shell script that will tell you every working IP device on your LAN. trunks.sh tells you all of the Asterisk trunks configured on your system. purgeCIDcache.sh will clean out the CallerID cache in the Asterisk database. convert2gsm.sh shows you how to convert a .wav file to .gsm. munin.pbx will install Munin on your system while awstats.pbx installs AWstats. s3cmd.faq tells you how to quickly activate the 
Don't forget to 
Support Issues. With any application as sophisticated as this one, you're bound to have questions. Blog comments are a terrible place to handle support issues although we welcome general comments about our articles and software. If you have particular support issues, we encourage you to get actively involved in the 
[
Twitter
