the definitive voip quick start guide

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The Definitive VoIP Quick Start Guide:

Introducing PBX in a Flash 2.0.6.3[1]

Whether youre just getting started with VoIP telephony or merely want to kick

the tires of PBX in a Flash 2.0.6.3.1, this guide is for you. Well try to cover the

basics as well as the fine points to get your PBX in a Flash system running on

almost any platform. Lets begin by telling you why PIAF is different and why it

matters. PBX in a Flash is the only Asterisk and FreePBX aggregation in

which most of the components are compiled as part of the install. PIAF is also

the only platform that lets you choose a stand-alone server platform, or a

turnkey virtual machine appliance for any Windows, Mac, Linux, or Solaris

desktop, or a hosted platform from RentPBX for only $15 a month! If youre cost

conscious or just want to tinker, weve got a simplified version with dozens of

utilities for the $35 Raspberry Pi.

Why does this matter? First, you get the very latest updates to the CentOS 6.3

operating system. Second, you get your choice of numerous Asterisk versions

including 1.8, 1.8-Certified, 10, and 11. Each is compiled from source on the fly.

Third, you get your choice of FreePBX versions as well: 2.8, 2.9., 2.10, or 2.11.

Fourth, you have a bloat-free platform that will let you easily add and compile

almost any Linux or Asterisk add-on in a matter of minutes but only if you need

it. One size fits all simply doesnt work for everybody in the VoIP world. Fifth,

you can adjust and fine-tune the existing PIAF setup to meet your own

requirements any time you like. This was especially handy last week when

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Digium announced that all prior versions of Asterisk had some major security

flaws. By tweaking a simple script, any PIAF server could be updated to the latest

version of Asterisk in under 30 minutes without losinganyof the existing

configuration. Of course youll also have access to the largest collection of free

Asterisk utilities and add-ons anywhere on the planet. If you dont need a

particular function, dont add it. If you do, its there for the taking on the PIAFForum [2] and can be installed in minutes. For the newbies that just want a

system that works, you can run the Incredible PBX script to generate a turnkey

system thats ready to plug in phones once you complete the installation. In less

than 5 minutes, youll have over 50 Asterisk applications with free Google Voice

calling in the U.S. and Canada. Last, but not least, we have the best VoIP

resource and collection of experts youll find anywhere on the PIAF Forum. And

its free!

Where to Begin?

Thinking back to when I began exploring VoIP telephony, I would suggest your

first order of business should be to have a seat in a quiet room and carefully

sketch out your objectives. Many want to test the waters privately before

recommending VoIP telephony to an employer. If thats your situation, then start

with the Incredible PBX 11 Virtual Machine [3] for VirtualBox. Once youve

downloaded the software, you can install it and be up and running on almost any

modern desktop computer in under 5 minutes. It also works with VMware [4]. We

have an airline using PBX in a Flash worldwide that started in just this way.

About half of our readers dont live in the United States. They live in 201 other

countries scattered around the globe (see our Flag Counter [5]). But theyre

unable to take advantage of Googles free phone service offering for calls in the

U.S. and Canada. Thats an integral part of all PIAF installs although you

certainly can use any SIP or IAX provider as well as traditional PSTN lines if you

want to. If thats your situation, take a careful look at the RentPBX Cloud

offering[6]. Theyre the best in the business and also one of our project sponsors.

For $15 a month, you can enjoy everything thats available to any U.S. resident

with a hosted PBX in a Flash or Incredible PBX account on one of their servers.


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Its a great way to take advantage of Google Voice and establish a corporate

presence in the U.S.

For those that just want a cheap VoIP solution with lots of bells and whistles for

your home or home office, a $35 Raspberry Pi running Incredible PBX [7] cant be

beat. You get almost the full feature set of Asterisk for the cost of an eveningmeal. Whats included? We could write a book about it. In fact, we almost have.

Start here [8]. Then go here [9].

Making a Hardware Selection

For the rest of you, the adventure begins here. Were going to assume that you

need a VoIP telephony solution that will support an office of up to several dozenemployees and that you have an Internet connection that will support whatever

your simultaneous call volume happens to be. This is above and beyond your

normal Internet traffic. To keep it simple, you need 100Kbps of bandwidth in

both directions for each call.1 And you need a router/firewall that can prioritize

VoIP traffic so that all your employees playing FarmVille wont cause

degradation in VoIP call quality. Almost any good home router can now provide

this functionality. Remember to disable ALG on your router, and its smooth

sailing.

For computer hardware, youll need a dedicated machine. There are many good

choices. Unless you have a burning desire to preserve your ties with Ma Bell, we

recommend limiting your Ma Bell lines to your main number. Most phone

companies can provide a service called multi-channel forwarding that lets

multiple inbound calls to your main number be routed to one or more VoIP DIDs


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much like companies do with 800-number calls. If this works for you, then any

good dual-core Atom computer will suffice. Youll find lots of suggestions in this

thread [10]. And the prices generally are in the $200-$400 range. For larger

companies and Asterisks capacity with beefier hardware, see these stress test

results [11].

If your requirements involve retention of dozens of Ma Bell lines and complex

routing of calls to multiple offices, then we would strongly recommend you

spend a couple thousand dollars with one of our consultants [12]. Theyre the

best in the business, and they do this for a living. They can easily save you the

cost of their services by guiding you through the hardware selection process.

They also have turnkey phone systems using much the same technology as youll

find in PBX in a Flash. You wont hurt our feelings.

Choosing the Right PIAF Platform

We get asked this question about a hundred times a week on the forums so here

goes. There are more than two dozen permutations and combinations of CentOS,

Asterisk, and FreePBX to choose from when you decide to deploy PBX in a Flash.

We always recommend the latest version of CentOS because it tends to be themost stable and also supports the most new hardware. You have a choice to make

between a 32-bit OS or 64-bit. Our preference is the 32-bit platform because it is

better supported. The performance difference is virtually unnoticeable for most

VoIP applications. With Asterisk, we always recommend an LTS release because

those have long-term support. That narrows your choices to Asterisk 1.8 or the

just released Asterisk 11. If you plan to use Digium Phones (and well get to

that), then youll want either Certified Asterisk 1.8 or Asterisk 11. The

conventional wisdom in the Asterisk community has been to avoid just released

Asterisk versions like the plague. We think weve turned the corner on that

approach. Asterisk 1.8 is close to end of life, and with Asterisk 11, youre in great

shape from a support standpoint for many years to come. We personally run

Asterisk 11 and have yet to find something that functionally would qualify as a

show stopper. Thats not to say there arent some bugs and security issues [13]

from time to time. A pretty serious collection of them was found just last week,


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but it affected all versions of Asterisk. So our bottom line is that Asterisk 11 is

the latest and greatest with the best feature set. If we were building a system for

a commercial business, it would be our hands-down choice. In the PBX in a Flash

world, we have colors for various versions of PBX in a Flash that support

different versions of Asterisk. Asterisk 11 happens to be PIAF-Green, Asterisk

1.8=PIAF-Purple, Asterisk 10=PIAF-Red, Certified Asterisk 1.8=PIAF-Brown.

Choosing the Right Phones

If there is one thing that will kill any new VoIP deployment, its choosing the

wrong phones. If you value your career, youll let that be an organization-driven

decision after carefully reviewing at least 6-12 phones that wont cause you daily

heartburn. You and your budget team can figure out the price points that work inyour organization keeping in mind that not everyone needs the same type of

telephone. Depending upon your staffing, the issue becomes how many different

phone sets are you and your colleagues capable of supporting and maintaining

on a long term basis.


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On February 1, Schmooze Com will release a public beta of their commercial End

Point Manager (EPM) at a price point of $25 per server. Theyve been using the

application internally to support their commercial customers for over a year so it

is not your typical beta software. Suffice it to say, its the best $25 you will ever

spend. You can sign up for an account with Schmooze through our commercial

support site [14] and purchase the software as soon as it becomes available. After

taking a look at the Admin User Guide [15], if youre a true pioneer, drop us a

note [16] and well get you a sneak peek. The beauty of this software is it gives

you the flexibility to support over 150 different VoIP phones [17] as well as other

devices almost effortlessly. Using a browser, you can configure and reconfigure

almost any phone on the market in a matter of minutes. So the question

becomes which phones should you show your business associates. That againshould be a decision by you and your management and budget teams, but collect

some information from end-users first. Choose a half dozen representative users

in your company and get each of them to fill out a questionnaire documenting

their 10 most frequent daily phone calls and listing each step of how they

processed those calls. That will give you a good idea about types and variety of

phones you need to consider for different groups of users. Cheaper rarely is

better. Theres a reason that everybody bought IBM Selectric typewriters. It had

nothing to do with cost. And phones can last a very long time, even lousy ones.

So dont blow it!

The phone brands that we would seriously consider include Cisco, Aastra, Snom,

Digium, Mitel, Polycom, Yealink, and Grandstream. Do you need BLF, call

parking or multiple line buttons, a hold button, conferencing, speakerphone, HD

voice, power over Ethernet support, distinctive ringtones for internal and various

types of external calls, Bluetooth, WiFi, web, SMS, or email access, an extranetwork port for a computer, headset support, customizable buttons (how

many?), quick dial keys, custom software, XML provisioning, VPN support? How

easy is it to transfer a call? Do you need to mimic key telephones? Also consider

color screens, touch screens, busy lamp indicators, extension modules (what

capacity?). What do we personally use: several Digium phones of various types, a

couple of Aastra phones, a Grandstream GXP2200, and a collection of Panasonic

cordless DECT phones, a fax machine, and Samsung Galaxy Note II connected


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through an OBi202 with an OBiBT Bluetooth Adapter to our PIAF server. Good

luck!

Installing PBX in a Flash

With the office politics out of the way, lets get to the fun stuff.

Download PIAF 2.0.6.3.1 ISO from SourceForge [18]

Burn the ISO to a USB Thumb Drive [19] or a DVD using a Mac [20] or Windows

machine [21]

Boot dedicated server using PIAF Flash Drive or DVD

For most deployments, choose the default install by pressingEnter.


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Leave the UTC System Clock option unchecked and pick your Time Zone. Tab to

OKand pressEnter.

Choose a very secureRoot Password. Tab to OKand pressEnter. Your server will

whir away for 5-10 minutes installing CentOS 6.3. When the reboot begins,

remove the DVD or USB thumb drive.

For today, were installing PBX in a Flash. So leave it highlighted, tab to OK, and

pressEnter.


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Now pick your PIAF flavor, tab to OK, and pressEnter.

ThePIAF Configuration Wizardwill load. PressEnterto begin.

Unlike any other aggregation, PIAF gives you the opportunity to fully configure

Asterisk usingmake menuconfigif you know what youre doing. For everyoneelse, type N and then confirm your choice.


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Next, youll need to choose your Time Zone again for PHP and FreePBX. Dont

worry if yours is missing. A newtimezone-setup utility is available in /root to

reconfigure this to any worldwide time zone.

Next, choose your version of FreePBX to install. Ignore the screen info regarding

Incredible PBX. Its out of date. The following limitations apply if you plan to

also install Incredible PBX and Incredible Fax:

Incredible PBX 3[22]requires PIAF-Purple and FreePBX 2.9

Incredible PBX 4[23]requires PIAF-Purple and FreePBX 2.10 (32-bit only)

Incredible PBX 11 will require PIAF-Green and FreePBX 2.11

Finally, you need to choose a very secure maintpassword for access to FreePBX

using a browser. You can pick your own, or the installer will generate one for you.

Dont forget it.


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The installer will give you one last chance to make changes. If everything looks

correct, press theEnterkey and go have lunch. Be sure you have a working

Internet connection to your server before you leave.

In about an hour, your server will reboot. You should be able to log in as root

using your root password. Write down the IP address of your server from the

status display (above) and verify that everything installed properly. Note that

Samba is disabled by default. If you want to use your server with Windows

Networking, run configure-samba once your server is up and running and youve

logged in.

Configuring PBX in a Flash


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Most PIAF Configuration is accomplished using the FreePBX Web GUI. Point

your browser to the IP address shown in the status display above to display your

PIAF Home Page. Click on the Users tab. ClickFreePBX Administration. When

prompted for your username and password, the username is maint. The password

will be the FreePBX master password you chose in the Config Module phase of

the PBX in a Flash installation procedure above.


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If youre new to Asterisk and FreePBX, heres the one paragraph primer on what

needs to happen before you can make free calls with Google Voice. Youll

obviously need a free Google Voice account. This gets you a phone number forpeople to call you and a vehicle to place calls to plain old telephones throughout

the U.S. and Canada at no cost. Youll also need a softphone or SIP phone to

actually place and receive calls. YATE makes a free softphone for PCs, Macs, and

Linux machines so download your favorite and install it on your desktop. Phones

connect to extensions in FreePBX to work with PBX in a Flash. Extensions talk to

trunks (like Google Voice) to make and receive calls. FreePBX uses outbound

routes to direct outgoing calls from extensions to trunks, and FreePBX uses

inbound routes to route incoming calls from trunks to extensions to make your

phones ring. In a nutshell, thats how a PBX works. There are lots of bells and

whistles that you can explore down the road.

To get a minimal system functioning to make and receive calls, heres the

2-minute drill. Youll need to set up at least one extension with voicemail, and

well configure a free Google Voice account for free calls in the U.S. and Canada.

Next, well set up inbound and outbound routes to manage incoming andoutgoing calls. Finally, well add a phone with your extension credentials.

A Few Words About Security. PBX in a Flash has been engineered to run on a

server sitting safely behind a hardware-based firewall with NO port exposure

from the Internet. Leave it that way! Its your wallet and phone bill that are at

stake. If youre running PBX in a Flash in a hosted environment with no

hardware-based firewall, then immediately read and heed our setup instructions

for Securing Your VoIP in the Cloud Server [24]. We would encourage you to visit

yourPIAF Home Page regularly. Its our primary way of alerting you to security

issues which arise. Youll see them posted (with links) in the RSS Feed shown

above. If you prefer, you can subscribe to the PIAF RSS Feed [25] or follow us on

Twitter [26]. For late-breaking enhancements, you also should regularly visit the

Bug Reporting & Fixes Topic [27] on the PIAF Forum.


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Extension Setup. Now lets 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 phones GUI to add bells and whistles. In FreePBX 2.10 or2.11, to create extension 201 (dont start with 200), clickApplications, 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 [randomly generated]

dtmfmode rfc2833

Voicemail Status Enabled

voicemail password 14332 [make this unique AND secure!]

email address [email protected] [if you want voicemail

messages emailed to you]

pager email address [email protected] [if you want to be paged

when voicemail messages arrive]

email attachment yes [if you want the voicemail message included in email]

play CID yes [if you want the CallerID played when you retrieve message]play envelope yes [if you want date/time of the message played before the

message]

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]


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vm context default

Write down the passwords. Youll 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 wont protect you from maliciousphone 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.

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 dont 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 dont pay anything

except when you actually use their service so you have nothing to lose.

For today, were going to take advantage of Googles current offer of free calling

in the U.S. and Canada through the end of 2013. You also get a free phone

number in your choice of area codes. PBX in a Flash now installs a Google Voice

module underFreePBX -> Connectivitythat lets you set up your Google Voice

account with PBX in a Flash in just a few seconds once you have your credentials.

A Word to the Wise: All good things come to an end especially those that are

free. So plan ahead with some alternate providers that keep your phones working

should Google decide to pull the plug or change the terms with Google Voice.

Signing Up for Google Voice. Youll need a dedicated Google Voice account to


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supportPBX 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!

Weve 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 account2, and use it

exclusively withPBX in a Flash. Google Voice no longer is by invitation only. If

youre in the U.S. or have a friend that is, head over to the Google Voice site [28]

and register. If youre living on another continent, see MisterQs posting[29] 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. Dont skip this step either.

Just enter the provided confirmation code when you tell Google to place the testcall to the phone number you entered. Once the number is registered, you can

disable it if youd 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. Thats the destination we need for

PBX in a Flash to function with Google Voice! Otherwise, inbound and/or

outbound calls will fail. If you dont 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 wont

work with PIAF.

While youre still in Google Voice [30] Settings, click on the Calls tab. Make sure

your settings match these:


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Call Screening OFF

Call Presentation OFF

Caller ID (In) Display Callers Number

Caller ID (Out) Dont Change Anything

Do Not Disturb OFF

Call Options (Enable Recording) OFFGlobal 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 Connectivity tab

in FreePBX 2.11 and choose Google Voice [Motif]. To Add a new Google Voice

account, just fill out the form. NOTE: The form has changed from prior releases

of FreePBX. Do NOT check the last box: Send Unanswered to GoogeVoice

Voicemail, or youll have problems receiving incoming calls.


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Google Voice Username is your Google Voice account name without @gmail.com.

Passwordis your Google Voice password. NOTE: Dont use 2-stage passwordprotection in this Google Voice account!Phone Numberis your 10-digit Google

Voice number. Next, check only the first two boxes:Add Trunk andAdd Outbound

Routes. Then click Submit Changes and reload FreePBX. Down the road, you can

add additional Google Voice numbers by clickingAdd GoogleVoice Accountoption

in the right margin and repeating the drill. For Google Apps support, see this

post [31] 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 dont 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, well

build a simple route that directs your Google Voice calls to extension 201.

Choose Connectivity -> Inbound Routes, leave all of the settings at their default

values except enter your 10-digit Google Voice number in theDID Numberfield.

Enable CallerID lookups by choosing CallerID Superfecta in the CID Lookup

Source pulldown. Then move to the Set Destination section and chooseExtensions

in the left pull-down and201 in the extension pull-down. Now click Submitand

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.


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Eliminating Audio and DTMF Problems. You can avoid one-way audio on calls

and touchtones that dont work with these simple settings in FreePBX: Settings

-> Asterisk SIP Settings. Just plug in your public IP address and your private IP

subnet. Then set ULAW as the only Audio Codec.

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. In FreePBX

2.11, chooseAdmin -> Module Admin and click on the Upgrade Notifications

shield on the right. Plug in your email address, click Submit, and save yourchanges. Done!

Setting Up a Desktop Softphone. PBX in a Flash supports all kinds of

telephones, but well start with the easy (free) one today. You can move on to

"real phones" once youre smitten with the VoIP bug. For today, youll need to

download a softphone to your desktop PC or Mac.

The easiest way to get started is to set up a YATE softphone on your Desktop

computer. Versions are available at no cost for Macs, PCs, and Linux machines.

Just download the appropriate one and install it from this link [32]. Once

installed, its a simple matter to plug in your extension credentials and start

making calls. Run the application and choose Settings -> Accounts and click the

New button. Fill in the blanks using the IP address of your server, 201 for your

account name, and whatever password you created for the extension. Click OK.


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Once you are registered to extension 201, close the Account window. Then click

on YATEs Telephony Tab and place your first call. Its that easy!

Monitoring Call Progress with Asterisk. That about covers the basics. Well

leave you with a tip on how to monitor whats happening with your PBX. There

are several good tools within the FreePBX GUI. Youll find them under the

Reports tab. In addition, Asterisk has its own Command Line Interface (CLI) that


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is accessible from the Linux command prompt. Just execute the following

command while logged in as root: asterisk -rvvvvvvvvvv.

Whats Next? Weve barely scratched the surface of what you can do with PBX in

a Flash. Log into your server as root and type help-pbxfor a list of simple install

scripts that can add almost any function you can imagine. And Incredible PBX 11is just around the corner. You can try it out in the Incredible PBX 11 Virtual

Machine [33] today.

[34]

Special Thanks. In just ten days, Nerd Vittles kicks off its 9th year of

publication. We wanted to express our thanks to all of you for visiting. In just the

last three months, weve again begun compiling a list of countries that are

represented in the Nerd Vittles Fan Club. Nerd Vittles has spread to 204

awesome countries so far. Just click on the Google Map and have a look for

yourself. Very inspiring. Thanks again for your support and Happy New Year!

Originally published[35]: Thursday, January 10, 2013


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[36]

Need help with Asterisk? Visit the PBX in a Flash Forum [37].

Or Try the New, Free PBX in a Flash Conference Bridge.

whos.amung.us If youre wondering what your fellow man is reading on Nerd

Vittles these days, wonder no more. Visit our new whos.amung.us [38] statistical

web site and check out whats happening. Its 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 from our special Vitelity sign-up link [39]. If youre 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

[40], the Nerd Vittles andPBX in a Flash projects get a few shekels down the road

while you get an incredible signup deal as well. The going rate for Vitelitys 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! ForPBX in a Flash users, heres a deal you cant (and shouldnt) refuse!

Sign up now, and you can purchase a Tier A DID with unlimited incoming calls

for just $3.99 a month. To check availability of local numbers and tiers of service


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http://nerdvittles.com/?p=46301.

http://www.pbxinaflash.com/community/index.php?forums/add-on-install-

instructions.10/

2.


https://sourceforge.net/projects/pbxinaflash/files/PIAF-2.0.6.3.1-VirtualMachine

/IncrediblePBX11.1/VMware/

4.

http://s08.flagcounter.com/more/QGz5.



from Vitelity, click here [41].Do not use this link to order your DIDs, or you wont

get the special pricing! Vitelitys 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,

of course, any balance is fully refundable if you decide to discontinue your

service.

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inShare [42]

There obviously are ways to dramatically reduce VoIP bandwidth. Heres a

table [43] that provides the details. The tradeoff is that youll need a beefier

computer to compress and decompress VoIP packets if you elect to use a

codec such as G.729. []

1.

You also can use a dedicated Google Apps account for Google Voice with the

latest version of the FreePBX module. Dont use your regular Google Apps

email address with Google Voice, or inbound calling will not work! []

2.


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http://pbxinaflash.com/community/index.php?threads/diy-piaf2-server-200.12460/10.

http://www.pbxinaflash.com/community/index.php?threads/stress-testing-

asterisk.11875/page-4

11.

http://pbxinaflash.com/support12.

http://www.pbxinaflash.com/community/index.php?threads/asterisk-security-

patches.14913/

13.

http://pbxinaflash.com/support/14.

http://literature.schmoozecom.com/end_point_manager-module/userguide

/End_Point_Manager-module.pdf

15.

http://pbxinaflash.com/about/comment.php16.

https://docs.google.com/viewer?url=http%3A%2F

%2Fliterature.schmoozecom.com%2Fend_point_manager-

module%2Fsupported_devices%2FEPM-Supported-Devices.pdf

17.

https://sourceforge.net/projects/pbxinaflash/files/18.


http://hints.macworld.com/article.php?story=2006061918101038920.

http://www.petri.co.il/how_to_write_iso_files_to_cd.htm21.




http://pbxinaflash.com/rssfeed.xml25.

http://twitter.com/NerdUno26.

http://pbxinaflash.com/community/index.php?forums/bug-reporting-and-fixes.9/27.

http://nerd.bz/dcYm6x28.

http://www.clazh.com/unlimited-free-international-calls-after-activating-google-voice-

outside-us/

29.

http://google.com/voice30.


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http://pbxinaflash.com/forum/showpost.php?p=70391&postcount=7431.

http://yateclient.yate.ro/32.


http://s08.flagcounter.com/gmap/QGz/34.


http://pbxinaflash.com/forum/36.

http://pbxinaflash.com/forum/37.

http://whos.amung.us/pro/stats/readers/zwh5/38.

http://pbxinaflash.com/vitelity/39.

http://pbxinaflash.com/vitelity/40.

http://www.vitelity.com/local_did_coverage/41.

javascript:void(0);42.

http://www.cisco.com/en/US/tech/tk652/tk698

/technologies_tech_note09186a0080094ae2.shtml

43.


the definitive voip quick start guide

Documents