Unlimited Free Calling with Google Voice

1. Unlimited Free Calling with Google Voice ............................................ 0

1. Introduction ................................................................................... 1
2. Free Calling from a Landline ............................................................. 1
3. Free Calling from a Cellphone (using the Voice plan) ........................... 2
4. Free Calling with Voice Over IP (VOIP) - Introduction........................... 6
5. The Gizmo5 Project - Background ..................................................... 6
6. Google Voice Integration with Gizmo5 ............................................... 7
7. Gizmo5 Integration with SIP Sorcery ................................................. 7
8. SIP Sorcery Integration with sipgate ................................................. 8
9. SIP Sorcery Scripting ...................................................................... 9
10. Unified Dial Plan ........................................................................... 9
11. Customizing the SIP Sorcery Unified Dialplan ...................................15
12. Connecting other SIP Clients and Devices ........................................16
13. Configuring and Testing your SIP Calling Chain ................................21
14. Additional setup screenshots ......................................................... 28
15. Revision History ........................................................................... 29

1. Introduction
Google Voice is a free service offered by Google and is designed to help you manage your
phone calls. When used for its original intended purpose, Google Voice can add unlimited
calling to and from US phone numbers to any landline. With a little more effort, that benefit
can be extended to cellphones with "calling circles", or plans that allow a certain number of
non-free phone numbers to be converted into free calls. With even more effort and a
suitable phone, the feature can be further extended to not only maintain free calling even
after a carrier decides those calls should no longer be free, but to also maintain free calling
while roaming outside of the United States. Finally, with the greatest amount of effort, one
or two lines with unlimited calling 24x7 can be added to any home or business.

2. Free Calling from a Landline

Google Voice offers two ways to enable 24x7 free inbound and outbound calling for landline
phones based in the US. First is a method that works the same way no matter what kind of
phone you end up using to make a call. From the main Google Voice web page
(http://voice.google.com) you are able to enter phone numbers or choose contacts from
your Google Voice contact list. With the phone number entered or selected, click the Call
button or link.

When viewed on a full PC browser, you will be asked which phone you would like Google
Voice to ring. When viewed on a mobile phone browser, Google Voice will dial the phone
number currently selected in the "My Mobile Number:" line on the Settings page. No matter
which version of the website you use, ensure that your landline is selected as the phone you
want to use for the call. Start the call and the selected phone will start ringing. As you don't
get the option to choose which phone to ring when using the mobile browser, you must
ensure you have the correct phone line selected prior to initiating the call.
Once Google Voice calls you, answer the phone. You will immediately hear the ringing sound
of an outbound call. This is Google Voice connecting you for free to the number you entered
or selected on the website. As landlines don't typically incur any charges to receive calls
(except "collect calls" where the recipient is asked to pay), there are no special steps to
take to receive calls for free. If you are home quite frequently, don't have unlimited long
distance calling on your house line, have access to the web on your cellphone, but don't
have a lot of cellular minutes to spare, this is a great way to make unlimited free long
distance calls from your home, without giving out your home phone number.

Just pull out you cellphone, bring up the Google Voice page, make sure your house phone is
selected as your Mobile phone number, and enter or select from the directory the phone
number you would like to call. This method works regardless of whether or not your Google
Voice number is a local call to your house number since all calls will be inbound to your
landline.

The second method is entirely different, and is most beneficial if the Google Voice number
you selected at signup is a local to you and you don't have long distance service. It is also
much simpler to use. Just pick up your phone and dial the Google Voice number you
selected. From the main menu, press the number 2 and you will be asked to dial the
number you wish to reach. Google Voice then connects you, for free, to the number you
dialed, so long as the number is in the US.

If your landline already has unlimited long distance or the numbers you call are within your
local or regional flat rate calling area, the main benefit of using Google Voice is to mask
your home phone. The people and businesses you call will see your Google Voice number on
their caller id, and will only need to know one phone number to reach you.

3. Free Calling from a Cellphone (using the Voice plan)

Central to Google Voice is the ability to manage your phone calls and SMS text messages.
Just as when using your landline, Google Voice can also help you mask your personal
cellphone number to other parties,showing them just one phone number for reaching you.
With a little planning (and a little cooperation from cellphone companies), Google Voice can
help you turn your cellphone into a flat-rate, unlimited use phone line. There are generally
two ways this can be achieved.

Cellphone usage is priced differently in the US from almost every other country. In general,
US cellphone users are charged minutes for both incoming and outgoing calls. The same is
true for text messages, although unlimited text messaging plans have been available across
all of the major carriers for many years. To help retain and build their customer bases over
the years, cellphone companies began by offering unlimited calling to other cellphones
operating on the same network. Even that wasn't even enough eventually, and companies
began allowing customers to select a handful of out-of-network phone numbers and
designate them as effectively being "in network". The out-of-network numbers don't have to
be other cellphones or landlines -- users can generally pick any set of numbers they like,
and they will never be charged for the minutes used during calls to or from the numbers on
the list.

As described above, there are two ways in which calls involving the Google Voice Number
selected by a Google Voice user can be used to branch out to free calls to any other phone
number in the US free of charge. First, the user can access the Google Voice website and
enter or select a number they want to call. Google Voice will then call the user at the phone
number where the user indicated they can be reached, showing a caller id of the user's own
Google Voice number. When the user answers the incoming call they will be connected to
the phone number entered or selected at the Google Voice website, free of charge.

Alternatively, if the user does not want to use the Google Voice website or does not have a
smartphone capable of browsing the web, the user may still dial their own Google Voice
number. Google Voice will recognize the call as coming from a phone that is managed by
the user, and will offer to play any waiting voicemail or connect them to another number.
The user may press 2 and then enter another phone number to which they will be
connected free of charge. It may become a hassle for users to dial phone numbers by hand
every time they want to make a phone call. Since their introduction, cellphones have always
had the ability to have keystroke sequences programmed into telephone numbers stored for
address book entries.

Two important programming codes available for numbers are the "pause" and the "wait".
When a pause is inserted into a phone number, the cellphone will dial the portion of the
phone number leading up to the pause. Then for each pause inserted into the stored
number, the phone will wait 2 seconds before continuing on to dial the remaining numbers.
When a wait is inserted into a code, the phone will again call the phone number stored
ahead of the wait code, but instead of waiting a certain amount of time prior to dialing the
remaining numbers, the phone will wait for the user to press a key to signify to the phone
that is okay to proceed in dialing the remaining numbers.

iPhone users who want to simplify dialing but weren't able to grab the official Google Voice
dialer have another option. There is a free app called Prefix Dialer. The app can be
configured to automatically prepend a dialing sequence in front of the numbers to be dialed
when calling an entry in the phone's address book. Once the app is installed, go into the
settings, and enter this special sequence:

Enter your Google Voice phone number, the letter "p", your pin, another letter "p", the
number 2, another "p", and optionally the number 1 (the final "1" seems to speed up dialing
a bit, but is not required).

For example, if your GV number is 555-666-7777, and your pin is 9999 then you would
enter "5556667777p9999p2p1". Then, whenever you want to place a call go into Prefix
Dialer & tap "select number". The app then shows you your normal iPhone contact list.
Select a contact and dial like you normally would. When the call is placed the app will dial
your prefix first, then the number you selected, thus routing the call through your Google
Voice account.

Separately, there are options for Google Voice users who don't have a smartphone (or a
data plan for their smartphone) but still want to enjoy free calling without typing 11 digits
for every outbound call. The solution is to update the phone numbers for each of their
contacts, starting with the most frequently called ones. Instead of just having the person's
phone number listed, the Google Voice user will update the number to read their own
Google Voice number, plus either a wait or a long enough series of pauses, the number 2,
another pause or two, and finally the number that was originally stored in that memory
location. This restores even "low tech" Google Voice users' ability to make unlimited free
outbound calls with little inconvenience.

It should be noted, however, that any frequently called numbers that are on the Google
Voice user's cellular network should not be changed. As Google Voice is not a cellphone
service, the parties receiving calls from new Google Voice users will now be charged
minutes to receive phones they have never been charged to receive before.

So by simply adding their own Google Voice number to their "Calling Circle", Google Voice
users have two options for making unlimited outbound calls without consuming any talk
minutes allocated to their account. One way requires a drastic change in the way in which
phone calls are placed by forcing users to visit a website instead of their own local address
book. The other way either requires a significant amount of extra button presses at the start
of each call or the systematic editing of every frequently called number in the user's address
book that is not a mobile phone on the user's own cellular network.

In addition, Google Voice has an option to further extend the benefits of this approach. By
going into the configuration settings of their Google Voice account, users are able to decide
what is shown on their caller ID when a call arrives through the Google Voice system. Users
may choose to see the actual caller ID presented by the caller, or simply mask all incoming
calls behind the Google Voice number. When calls arrive on the user's Google Voice line,
calls presented to the user with the user's own Google Voice number as the caller ID will
cause the user's carrier to recognize that number as being in their calling circle and
therefore free.

There is a third way in which users can enjoy unlimited calling. On select smartphones,
particularly those running Google's Android operating system and RIM's Blackberry phones,
Google has released software that simplifies outbound dialing. The user brings up an
alternate dialer, which still typically has access to the user's local address book stored on
their smartphone. When the user chooses a number to dial, the alternate dialer performs
two steps in parallel. First, the dialer uses the phone's Internet connection to notify Google
Voice's servers that a call is about to arrive from the user's cellphone and the message
includes the phone number they are actually trying to reach. Then the dialer triggers the
phone to call an entirely different, unpublished number to log into the Google Voice system.
Once the Google Voice system has processed the message sent previously by the alternate
dialer, the system answers the inbound call coming from the user's cellphone and connects
the caller to the number they originally specified in the alternate dialer.

This is significant because there's no way for users to know which Google Voice "access
number" the user's handset is going to call and yet, this number has to be detected by the
user and added to the user's calling circle. If the user does not figure out this access
number and add it to their calling circle, they will be under the misconception that
all of their outbound calls are free, when in fact they are not.

When Google Voice users with calling circles add their Google Voice number (and the access
numbers used by smartphones) to their calling circle and choose to have Google Voice
display their own Google Voice number as the caller ID on incoming calls, they are now able
to make and receive unlimited calls with their cellphones under certain conditions:

1) The cellphone is on their carrier's home network (i.e. not roaming),

2) The user always uses a dialing method that routes calls through the Google Voice servers
and all entrance numbers are listed in their calling circle,

3) Parties calling the user call them through their Google Voice number,

4) The user maintains at least the minimum calling plan that qualifies for calling circles
Users should note that the access number dialed by the alternate dialer on Android and
Blackberry phones is subject to change at any time, and it is the user's responsibility to
ensure that the access number being dialed matches the number on their calling circle for
that purpose. The alternate dialer on Blackberry phones does not entirely mask the calling
process from the user. Once the dialer transmits the target phone number to the Google
Voice servers over the Internet, the dialer app initiates a standard phone call, allowing the
user to see the standard screen displayed while a call is in progress.

With this in mind, a common suggestion given to Blackberry users is to create an entry in
the phone's local address book for Google Voice. The two numbers stored in that entry
would be the user's own personal Google Voice number as well as the access number they
detect as being called by the alternate dialer. The user may also go as far as adding a
picture to the entry, such as this one found here:

http://www.iphonefreak.com/wp-content/uploads/2009/06/
gv_mobile_logo_new_250x250.jpg

Doing so makes it very easy for Blackberry users to recognize when the access number has
changed since the "Call In Progress" screen will simply show a phone number instead of the
name "Google Voice" and the icon displayed above will not be shown. When the icon and
name are not displayed on outbound calls initiated by the alternate dialer, the user will
know that the access number being dialed by the dialer app is not on their calling circle and
the outbound calls will not be free of charge until the new number is added to the calling
circle.

The alternate dialer on Android phones apparently mask the entire process of dialing the
access number, therefore making it a bit more difficult to identify the access number being
dialed. Furthermore, adding that number and the caller ID picture above to an entry created
for Google Voice is of no value since the standard "Call In Progress" screen is not displayed.
4. Free Calling with Voice Over IP (VOIP) - Introduction
Voice Over Internet Protocol (Voice Over IP, or simply "VOIP") is a generic term that applies
to methods and approaches for allowing users to transmit audio over Internet connections.
There are many approaches and technologies applied to implement VOIP, and a few
standards have been developed to ease the interaction between two or more systems
looking to establish a session. One of the most popular approaches is called Session
Initiation Protocol, or SIP. SIP is able to establish connections covering a variety of
communications, but for for the purpose of this discussion, SIP should be thought of as a
way to link two or more systems together for the purpose of establishing, maintaining, and
terminating phone calls. Generally speaking, a SIP server is a system that either connects
to other systems or to end users, either over SIP or through a gateway function that links
"digital" phone calls to telephone devices.

What end users tend to think of as a "phone line" can be hosted by a SIP server and lines
can be activated or deactivated by an authorized person identified through their login
credentials. Just like how people login to Google's properties to check their Gmail account
for new email, see their custom dashboard in iGoogle, chat with their friends and family on
Google Talk or read the headlines in Google Reader, users of SIP lines can bring up phone
lines by using special software to authenticate against a server using unique credentials.
When the SIP phone line is activated by the authorized user, phone calls destined for that
phone line are presented to the end point the user has designated as their personal
interface to the phone line.

Endpoints may take the form of software on a laptop or PC that looks just like a desktop
telephone, software that runs on small, handheld computers the size of a cellphone,
software that runs on an actual cellphone or (with the help of another device) a standard
telephone sitting right on their desk. Without getting too far into details, it is just important
to know that what we know as "phone calls" can now freely hop back and forth between the
old fashioned wire technology you see in your walls behind the phone plug and computers
communicating with each other over the Internet. That's how the phone companies have
done it for years, and you are free to join in the fun.

Since long before GrandCentral (the predecessor to Google Voice) hit the scene, services
had been popping up, offering low cost phone calls with very low per-minute rates,
particularly for international callers. Some companies allowed customers to access their
system via regular telephone lines while others only allowed calls to traverse their network
in digital form. Since Google Voice offers publicly accessible phone lines and uses high tech
approaches to routing and managing phone calls on those lines, its only appropriate that
Google would allow a pure IT-based approach to allowing users to connect to their Google
Voice phone lines in the same way. And Google did allow this for a short time.

5. The Gizmo5 Project - Background

After a brief period of allowing all users direct access to their Google Voice phone lines via
SIP, Google restricted the purely digital access through an established Internet phone line
provider, Gizmo5. Now Gizmo5 had its own digital phone line service offering, complete with
their own domestic and international calling rates. But by accepting digital phone calls from
Google Voice and allowing Google Voice users to receive phone calls from Google Voice
completely free of charge, their revenue model was threatened. Remember, when you enter
or select a phone number to call on the Google Voice website, Google Voice calls you first
and when you answer, connects you to the other party.

So if Gizmo5 was granting free access to end users to free digital-only phone lines but also
had the only all-digital path to Google Voice, they were destined to receive a stampede of
new Google Voice users who had no interest in using their chargeable services.
Furthermore, since Gizmo5 offers free forwarding of inbound calls to other destinations over
SIP, this meant that anyone worldwide could suddenly sign up for free services from Gizmo5
while waiting for a free phone line from Google Voice. Google worked very hard to limit the
earliest eligible beta testers to those located in the US, but Gizmo5 breaks that requirement
and will forward your phone calls anywhere in the world over SIP.

After a several months of beta testing, Google recognized that not only does Gizmo5 play a
key role in linking some of Google's users with a very popular product, the popularity of the
product paired up with Gizmo5's generosity was quickly eating away at their bottom line.
Instead watching Gizmo5 collapse into itself, taking a successful, established business with
it as well as thousands of Google users using Gizmo5's services for free, Google bought
Gizmo5 and is currently in the process of rebranding their product line and apparently
preparing to better integrate the service with Google's offerings.

6. Google Voice Integration with Gizmo5

After a torrid build-up of interest in Gizmo5 as the only provider with SIP access to Google
Voice, Gizmo5 introduced some integration with Google Voice. Recognizing that Google
Voice directly competes with and undercuts their long distance calling product, Gizmo5
added some configuration settings to the user accounts, allowing users to specify their
Google Voice credentials. Then, users would initiate an outbound call through Gizmo5's own
systems, not Google's. The Gizmo5 servers would then interact with the Google Voice
servers to initiate an outbound call but connect the call to the Gizmo5 user who initiated the
call. Because of the direct competition, Gizmo5 limited calls inititated in this way to 3
minutes in length.

Gizmo5 did not limit the length of calls that arrived inbound from the Google Voice servers,
however. This means that all calls arriving into Gizmo5 accounts from Google Voice are
unlimited in length. Recall that when Google Voice users initiate outbound calls by entering
or selecting phone numbers at the web site, Google Voice always proceeds by calling the
user first at a number they specify, only to connect the user to the desired phone number
when they answer the call. From the very beginning, Google Voice users with Gizmo5
accounts have been able to make calls of unlimited length over SIP merely by initiating the
calls at the Google Voice website.

Since SIP is a computer-to-computer communication system, this means that users with
devices capable of SIP communication can now enjoy free unlimited calling by
authenticating their SIP client against their SIP account at Gizmo5 and initiating outbound
calls via the Google Voice website and indicating to Google Voice that they can be reached
through their Gizmo5 phone line. This also means that devices that have not historically
been thought of as telephones can now serve their users in that role.

7. Gizmo5 Integration with SIP Sorcery

Among the many SIP phone line providers on the Internet, SIP Sorcery presents an
interesting end user experience. SIP Sorcery was developed to offer a flexible way to control
the flow of phone calls while in their digital form. What is unique about SIP Sorcery's
approach is that they expose a complete programmable environment, allowing skilled users
to develop robust and complex schemes for routing calls between networks. When
GrandCentral reopened its doors as Google Voice and Google Voice restricted its SIP-based
communications to Gizmo5, SIP Sorcery seized a unique opportunity.

See, Gizmo5 still offers to forward all inbound calls to another destination free of charge, so
long as the connection is via SIP. This is of no consequence to SIP Sorcery because they
only deal in SIP links. SIP Sorcery unilaterally updated their system to include support for
Google Voice. They developed and made available a script function that uses Google Voice
to establish phone calls which are then presented via SIP to SIP Sorcery and on to their
connected users. SIP Sorcery uses the phone numbers entered via a SIP client to kick off
the script that establishes a call at Google Voice.

What does all this mean? It means that users who connect to SIP Sorcery via SIP are
able to dial phone numbers directly in their SIP clients or on their analog phones
attached to a SIP-compatible phone adapter. SIP Sorcery then uses Google Voice
to place unlimited free calls to any US phone number entered by the user.

It also means, by extension, that users who connect their SIP clients directly to
Gizmo5 or other SIP gateway providers and don't want to pay for their value
added services are not utilizing the service in the most efficient and cost effective
way.

8. SIP Sorcery Integration with sipgate

SIP Sorcery is not only able to process and route SIP calls, but is also able to serve as a SIP
client to SIP providers. This capability enables SIP Sorcery users to use any other telecom
provider that offers SIP access to their users' phone lines. Sipgate.com is one such provider.
In order to enable 24x7 free calling for Google Voice and sipgate users using just a SIP
client, follow these simple steps to use the dialplan in the next section.

Note: These instructions will be sipgate-specific until Google re-opens Gizmo5 for
signups. If you have a Gizmo5 account already and would like to know how to
enable free calling without the 3-minute limit, please ask in the forum until the
new process is known.

1. Sign up for a free Residential Sipgate ONE account at sipgate.com. Go into your settings
page and take note of both your regular phone number (usually in area code 415) as well as
your SIP credentials.

2. Sign up for a free account at SIPSorcery.com. You will need to install Microsoft Silverlight
in your browser to perform this step and complete the rest of this process.

3. On the SIP Providers page, add a new provider named "sipgate" and enter your SIP
username and password provided by sipgate. Enter sipgate.com as the server. Click "show
advanced settings" and add sipgate.com as the proxy. Check the "Register" checkbox and
make sure that the registered contact shows "sip:" followed your SIP Sorcery name and
"@sipsorcery.com". Click Add when you have finished. After a few seconds, sipgate should
automatically appear in the "SIP Provider Registrations" list at the bottom of the screen. If it
does not, double check the SIP credentials and server settings against those shown on your
sipgate account settings page.
4. Configure your SIP client, SIP-compatible softphone or SIP-compatible ATA to login to
your sipsorcery account. Check the "SIP Accounts" page of your SIP Sorcery account and
make sure an entry appears in the "SIP Bindings" list at the bottom of the page.

5. Use a phone to call your sipgate number and make sure the phone attached to your SIP
Sorcery account rings. Answer the call and make sure you can hear sound in both
directions.

6. Login to Google Voice, click Settings, Phones, and add your sipgate phone number as a
new managed phone. Google Voice will display a two digit number on your screen. Click
Connect and Google Voice will call your sipgate number and ask you to enter the code
displayed. Once you successfully add the new line, make sure it is checked on your Google
Voice phone list.

Note: You are now ready to receive inbound calls to your Google Voice number for free. You
may also enter phone numbers on the Google Voice website and select your sipgate number
as the phone you wish to use to make free outbound calls. There is one last step required to
enable 24x7 free unlimited outbound dialing from your SIP-compatible softphone or
standard telephone attached to your SIP-compatible ATA.

7. Copy the SIP Sorcery dialplan below, go to the Dial Plans page in SIP Sorcery, click on
default, and paste the dialplan, overwriting the short default script already there. Customize
the portions of the dialplan highlighted in blue below according to section 11. The script will
not function correctly and you will not be able to make free outbound calls until you
customize all of the highlighted text.

If you've made it here, you should be all set. If you followed all the steps properly, you
should be able to start your SIP client or pick up your ATA-connected phone and dial any
number in the US for free. If you are unfamiliar with other SIP clients and SIP-compatible
devices available to you, check out Section 12.

NOTE: If you are unable to follow these steps and just can't seem to get outbound dialing to
work, please see section 13 of this document for more detailed instructions, including
screenshots of the setup process.

9. SIP Sorcery Scripting

Key to taking advantage of SIP Sorcery's capabilities is their scripting feature. It is
important to remember in all this that SIP is just a method for enabling computers to
exchange audio (and other information) in a way that is compatible with what people know
as "telephone calls". The scripts that SIP Sorcery supports are called dialplans. Dialpans
instruct SIP Sorcery on how to route calls based on a variety of parameters. Only outbound
dialplans are needed for integration with Google Voice.

10. Unified Dial Plan

Here is a dialplan adaptable to both sipgate and Gizmo5 that offers a speed dial directory
that is accessible from every attached phone. It is also capable of automatic routing across
multiple SIP providers based on the phone number you are calling, also developed by
MikeTelis at mysipswitch.com. You will need to edit the Google Voice section with your
Google Voice username, password, and your assigned sipgate number.
For the code below - entries highlighted in blue need to be modified before saving the
dialplan. There are 5 entries that need to be customized and 2 highlighted hash marks in
each of two sections, one of which in each section MUST be removed in order for dialing to
function.

#============================= BEGIN CODE

======================#

#Ruby

# Note: This script works with both gizmo5 and sipgate but some
# EDITS ARE REQUIRED to indicate which SIP provider is being used

# ---------------BEGIN MANDATORY ENTRIES SECTION-----------------------#

Area = '217' # my area code, this will be added to 7-digit dialouts

Tz = -6 # my time zone (GMT format, e.g. Central = -6)

# ---- Enter your Google Voice ("GV") and Sipgate account credentials----#

GV_USER = "user@gmail.com" # my GV e-mail address (user@gmail.com)

GV_PASS = "password" # my GV password
SIP_NUMBER = "xxxyyyzzzz" # my 10-digit Sipgate or Gizmo5 number

# ***************Google Voice Configuration Section *********************

# You must remove the "#" from the start of the one of the two lines in this section
# that correlates to your provider.
# Dialing WILL NOT WORK UNTIL YOU REMOVE THE "#" FROM THE APPROPRIATE LINE

def googleVoice

# sys.GoogleVoiceCall(GV_USER,GV_PASS,SIP_NUMBER,@num,'.*',1,30) # sipgate
# sys.GoogleVoiceCall(GV_USER,GV_PASS,SIP_NUMBER,@num,'.*',7,30) # Gizmo5

end

# ***************SIP Provider Configuration Section ********************

# You must remove the "#" from the start of one of the two lines in this section
# that correlates to your provider.
# Select the appropriate line starting with '5' and remove the first '#'
# Dialing WILL NOT WORK UNTIL YOU REMOVE THE "#" FROM THE APPROPRIATE LINE

VSPtab = {
'0' => '00 @ F9', # Future-nine default route
'2' => '02 @ F9', # Future-nine grey route
'3' => '03 @ F9', # Future-nine white route
'4' => '04 @ F9', # Future-nine premium route
# '5' => '@ sipgate', # sipgate
# '5' => '@ gizmo', # gizmo

# -----------------END MANDATORY ENTRIES SECTION--------------------#

# --------- BEGIN OPTIONAL SECTION FOR SPEED DIAL -------#
# Speed dial entries. Format: "short code" => "destination (POTS or SIP)"

Speeddial = {
'0' => '123456789', # my 0 (zero) key's speeddial number
'411' => '8004664411', # Google's Directory Assistance, GOOG-411
'303' => '303@sip.blueface.ie', # Blueface speaking clock (Ireland time)
'266' => '4153767253@podlinez.net', # CNN Headlines (266 = "CNN")
'677' => '8186882773@podlinez.net', # NPR's most e-mailed stories (677
="NPR")
'742' => '6506441934@podlinez.net', # Prairie Home Companion's, or PHC's
# News from Lake Wobegon (742 = "PHC")
'932' => '8009328437', # 800-WEATHERCALL (9328437 =
"WEA[ther]")

}
# ---------- END OPTIONAL SPEED DIAL ENTRY SECTION ----------------#

# Serviced domains, must be in lowercase!

Domains = ['sipsorcery.com','174.129.236.7']

# Excluded Prefixes. Provides a safeguard against accidentally calling premium

# numbers. Accepts both strings and patterns, spaces ignored

ExcludedPrefixes = [
' 1 (900 | 976 | 809)', # USA Premium
'44 (9 | 55 | 87 (0|1))', # UK Premium
'44 84 (4|5)', # UK Local Premium
'44 70', # UK Personal Premium
'43 (8|9)', # Austria Premium
'32 (7|90)', # Belgium Premium
'45 (1 | 50 (1|2|3) | 70 (1|2))', # Denmark Premium
'45 (8|9) 0', # Denmark Premium (...)
'33 (7|9)', # France Premium
'49 (1 | 900)', # Germany Premium
'39 [^3]', # Italy Premium (...)
'31 (14 | 6 (3|8|9) | 8 | 9)', # Netherlands Premium (...)
'48 (39 | (2|7|8) 0)', # Poland Premium
'46 9 (00 | 39 | 44)', # Sweden Premium
'41 90 (0|1|6)', # Switzerland Premium
]

# ******************** s e l e c t V S P *******************************

def selectVSP # VoIP provider selection

# Reject calls to premium numbers unless VSP was forced

ExcludedPrefixes.each {|p| p.gsub!(/\s*/,''); sys.Respond(503,"Calls to #{$1}* not

allowed") if @num =~ Regexp.new("^(#{p})")}

case @num
when /^1([2-9]\d\d)/ # North America

case $1 # check area code

when "808", "907", "867" # AK, HI, Canada Yukon calls are now free with GV
googleVoice # route(4,"Destination - Alaska, Hawaii, Yukon")

when /(800|866|877|888|415)/ # US toll-free calls

googleVoice # route(5,"Destination - US toll-free or sipgate")

else
googleVoice
sys.Log("GoogleVoiceCall failed, routing thru F9")
route(4,"Destination - North America")
end

when /^972(5|6)/ # Israel mobile

route(3,"Destination - Israel mobile")
else
route(0,"Default route applied")
end
end

# ************************** C A L L S W I T C H **********************

def callswitch(num,*args)
route # Initialize vars

@num = num unless @num = Speeddial[num] # If there is speed dial entry for it...

@l = "URI dialing: #{@num}" # Assume URI dialing

unless @num =~ /@/ # If we already have URI, skip all number processing
@num.gsub!(/%(..)/) {$1.to_i(16).chr} # Convert %hh into ASCII

#==========================================================
# Note: this section is important to understand because it lets users choose a specific
# SIP Provider for each outbound call. Perfect for users both sipgate and G5
# accounts who want to be able to choose which provider to use. Let's develop
# this section to make the script portable to both G5 and sipgate users as well
# as user with other 3rd party SIP providers

if @num =~ /^#(.)(.*)/ # If number starts with '#'

@p = $1; @num = $2 # next char is VSP code
end
#===========================================================

@num.gsub!(/[^0-9*+]/,'') # Delete all fancy chars (only digits, '+' and '*' allowed)

# sub! below removes prefixes:

# '+' - international format
# 00 - European style international prefix (00)
 # 011 - US style international prefix (011)

unless @num.sub!(/^(\+|00|011)/,'') # If no international prefix, process special cases

below
case @num
when /^[2-9]\d{6,6}$/ # Local call, 7-digit number
@num = "1#{Area}#{@num}" # prefix it with country and area code
when /^[01]?([2-9]\d{9,9})/ # US number with or without "1" country code
@num = '1' + $1 # add country code and truncate number to 10-digit
when /^\*/ # Voxalot voicemail, echotest & other special numbers
else
sys.Respond(603,'Wrong number, check & dial again')
end
end

sys.Log("Number in ENUM format: #{@num}")

@l = "Forced routing to provider #{@p}, template '#{VSPtab[@p]}'" # Assume user

explicitly selected VSP

if @p.empty? # Automatic VSP selection?

# Invoke selectVSP prior to ENUM lookup just in case we need to modify @num

route # re-initialize variables

selectVSP # Pick appropriate provider for the call

if enumuri = sys.ENUMLookup("+#{@num}.e164.org") # Check if NAPTR exists for the

number
sys.Log("ENUM entry found: '#{enumuri}'") # If yes, call that URI
sys.Dial(enumuri) # if call failed, call via regular VSP.
status() # If this is not what you want, add "return"
sys.Log("Call to #{enumuri} failed (#{@reason}), will call again via regular VoIP
provider")
end

end # @p.empty
end # URI

dial(*args) # dial selected number or URI

end

# ******************************* D I A L ********************************

def dial(*args)
sys.Log(@l) unless @l.empty? # for the record :)
if tpl=VSPtab[@p.to_s] # if provider is in our table
tpl.gsub!(/\s*/,'') # Remove spaces
@num = tpl.gsub(/@/,@num+'@') # Insert number before '@'
end
sys.Dial(@num,*args) # Dial
status() # We shouldn't be here! Get error code...
sys.Log("Call failed: code #{@code}, #{@reason}")
end
# ****************************** R O U T E *******************************

def route(p='', l='')

@p = p; @l = l
end

# ***************************** S T A T U S ******************************

def status
if (ptr = sys.LastDialled[0]).nil?
@code = 487; @reason = 'Cancelled by Sipsorcery'
else
ptr = ptr.TransactionFinalResponse
@code = ptr.StatusCode; @reason = ptr.ReasonPhrase
# sys.Log("#{ptr.ToString()}")
end
end

# ******************************* M A I N *****************************
begin
sys.Log("** Call from #{req.Header.From.ToString()} to #{req.URI.User} **")

t = Time.now + ((Tz+8)*60*60) # Get current time and adjust to local. SS Server is in

GMT-8
sys.Log(t.strftime('Local time: %c'))

if sys.In # If incoming call...

name = req.Header.from.FromURI.User.to_s # get a copy of caller's number

# Prefix 10-digit numbers with "1" (US country code).

# Some DID send Caller ID without country code

name = ('1' + name) if name =~ /^[2-9]\d\d[2-9]\d{6}$/

name.sub!(/^011/,'') # Remove 011 prefix added by Google Voice

sys.Log("FromName: '#{name}'")
req.Header.From.FromName = name # Set FromName for sys.Dial
# Change FromURI.User, too - or else Bria won't find contact in its phonebook!
req.Header.from.FromURI.User = name

if sys.IsAvailable() # If my ATA is registered

callswitch("#{sys.Username}@local") # forward all incoming calls to it
elsif (8..23) === t.hour # else forward calls to my home
sys.Log("#{sys.Username} is not online, forwarding call to home number...")
callswitch("0",35) # Note that '0' is in my speed dial list
end

sys.Respond(480, "#{sys.Username} Not online") # switch to voice mail

else # Outbound call ...

# check if it's URI or phone number.

 # If destination's host is in our domain, it's a phone call

num = req.URI.User.to_s; reqHost = req.URI.Host.to_s # Get User and Host

host = reqHost.downcase.slice(/[^:]+/) # Convert to lowercase and delete
optional ":port"

num << '@' << reqHost unless Domains.find {|x| x == host} # URI dialing unless host
is in our domain list

callswitch(num)

end
sys.Respond(@code,@reason) # Forward error code to ATA
rescue
# Gives a lot more details at what went wrong (borrowed from Myatus' dialplan)
sys.Log("** Error: " + $!) unless $!.to_s =~ /Thread was being aborted./
end

#============================= END CODE

======================#

11. Customizing the SIP Sorcery Unified Dialplan

The very top of the script contains the "Mandatory Entries" section. First, there are 5 pieces
of information highlighted in blue that you will need to provide. The five details are

1) Your local area code. This restores 7-digit dialing to Google Voice in your home area
code.
2) Your time zone.
3) Your Google account in full email address form (typically "user@gmail.com")
4) Your Google account's password
5) The 10-digit phone number of the SIP provider you plan to use. For Gizmo5 users, this
will be a number in area code 747. For sipgate users this must be the number that was
assigned to you by sipgate. It can be in almost any area code, but more often than not, it is
from area code 415.

NEW STEP: Once you have entered those five details, you must review the next two
sections of code. Each section will have two lines with leading hash marks ("#"), and you
must determine which ONE line in each section must have its leading hash mark removed.
It is easy to determine which line in each section needs to be edited based on the provider
you decided to use.

After the mandatory section is the optional speed dial directory. This function has two
benefits. First, it brings customizable "short codes" to standard phones. This allows numeric
codes to be dialed and have those codes interpreted to trigger a connection to a specific
destination. For example, a user can create "extensions". So for instance, 411 is translated
to 1-800-466-4411, which is Google's 411 (information by phone) service. (Note that
4664411 spells "GOOG-411" on a standard telephone keypad.)

IMPORTANT: If you are disconnecting your home phone service and do not have a
cellphone, be advised that Google Voice does not provide 911 service. You should take a
minute to call your emergency services center ON A NON-EMERGENCY NUMBER and ask
them for the best alternative number to call if you don't have 911 service.

URGENT NOTE: It is always best to call 911 from a live phone line provided by the phone
company. Your second best option is your cellphone, preferably one with GPS functionality.
Your very last option should be to use Google Voice to call the alternate number provided to
you by your emergency services center. They will not have automatic addresses or GPS
coordinates, so they will have the least amount of information available if you call during an
emergency. They will also typically answer 911 calls first so you may not get an immediate
response when it is most urgently needed.

A popular use for the speed dial directory is to listen to podcasts collected by podcast
aggregator Podlinez. They collect thousands of regularly updated podcasts and link them to
phone numbers. By calling the assigned phone number, the user will immediately hear the
latest episode of the associated podcast. Podlinez takes the extra step, however, of making
each and every phone line also available via SIP. This means that SIP Sorcery can be
configured to call any of those numbers directly over SIP instead of using Google Voice to
call into Podlinez via a standard phone line. And since the speed dial directory supports
translating short codes to other SIP-accessible phone lines, users are free to choose any
number of podcasts from the Podlinez directory and add them to their speed dial list.

For example, the CNN Radio top stories podcast is updated hourly and available to all
phones at 1-415-376-7253. Podlinez also makes that phone number available via SIP at
"4153767253@podlinez.net". Instead of dialing the entire number or creating an entry in
the address book for each device attached to your SIP Sorcery account, a "short code"
number can be added to the custom speed dial directory at the top of the script. As soon as
the script has been updated, all attached devices can dial the short code and hear the
headlines. Since 266 spells "CNN" on a standard telephone keypad, the short code can be
added to the speed dial directory with this simple line placed in the middle of the directory:

'266' => '4153767253@podlinez.net', # CNN Headlines (266 = "CNN")

Again, all phone numbers listed in the Podlinez directory can be added to the speed dial
directory by creating an entry formatted like the one above. This is not required as users
are still free to call the traditional phone lines to hear the podcasts, but those lines are
occasionally busy. Accessing the lines via SIP bypasses the traditional phone lines, leaving
them open for other users who are not SIP enabled, and also eliminating the chance of
calling at a time when the phone lines are simply out of capacity.

12. Connecting other SIP Clients and Devices

The hard part is out of the way! You are now free to connect any SIP client software or SIP-
compatible devices to your SIP Sorcery account and enjoy unlimited calling to and from US-
based phone numbers. SIP client software is available on many platforms. These include but
are not limited to:

• sipdroid for use on Android cellphones and Internet tablets

• iSip for the iPhone and iPod Touch
• Internet Call for Maemo devices such as the Nokia N810 Internet Tablet
• SIP Phone (note: Get the real name!) on Symbian cellphones

Many SIP-compatible Analog Telephone Adapters (ATAs) are also able to login to your SIP
Sorcery account. One popular device is the Linksys PAP2T-NA ATA.
There's a wikipedia overview of the PAP2T-NA, along with some useful links located here:
http://en.wikipedia.org/wiki/Linksys_PAP2

The support page, including a link for downloading the latest firmware, for the PAP2T-NA
may be found here:
http://www.cisco.com/en/US/products/ps10029/index.html

A PDF version of the Admin Guide may be found here:

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/
ATA_AG_v3_NC-WEB.pdf

Following are screenshots and instructions for configuring your PAP2-NA to authenticate to
your SIP Sorcery account.

1. Click "Line 1" from the Administration page

2. Enter your SIP Sorcery credentials.

That's it. Hook up a standard analog phone to the Line 1 port on your Linksys device, pick
up the phone, and enjoy unlimited calling to and from US-based phone numbers.

If you would like an alternative approach to configuring your Linksys ATA, see the
configuration wizard available at Voxilla, available here:

http://voxilla.com/voxilla/tools/device-configuration-wizard/linksys-pap2-pap2t-
configuration-wizard

1. Connect the Linksys device to an open port on your home router. Attach an analog phone
to line 1. Pick up the phone and dial "****" (Four stars). At the voice prompt, dial "110#"
and take down the ATA's IP address that is given to you.

2. Using the Voxilla configuration utility, enter the IP address of the router in Step 1.
Choose "SIPphone (STUN)" in Step 2.
3. Follow these examples for completing steps 3 through 6.

4. For steps 7 through 10, skip the Dial Plan, choose Line 1, select your timezone, and then
start the process to configure your adapter.
5. When the Linksys configuration screen appears, change the proxy server from
proxy01.sipphone.com to sipgate.com. If the screen does not look like this example, make
sure you are on the Admin page, not the User page.
6. In case of irrecoverable errors, and you need to reset the Linksys adapter to the factory
state, try these instructions found here:

http://forums.linksysbycisco.com/linksys/board/
message?board.id=VoIP_Adapters&message.id=4659

A. Disconnect the ethernet cable from your PAP2T-NA

B. Dial ****73738# and wait for the Interactive Voice Menu to get activated
(Note: "73738" spells "RESET" on your phone's dialpad)

If asked for a password, try 8995523# or 7756112# or 5465866# or 50274537# or

78196365#
One of those passwords should work for you)

C. Enter 1 to confirm, press #, and then hang up the phone.

13. Configuring and Testing your SIP Calling Chain

If you are unable to follow the instructions in Section 8, following are screenshots and
descriptions of the steps required to sign up for free accounts at various websites and
instructions for connecting it all together for free calling. If you have completed all of the
steps leading up to this point and are able to place calls from SIP clients or your ATA, you
are finished and do not need to read this section.

1. Go to sipgate.com and click the "Sign up" link in the menu bar
2. From the signup page, enter your cellphone number and choose your carrier.
3. Sipgate will send a 4-digit code to your cellphone in a text message and wait for you to
enter it on the website. Enter the code in the appropriate field. (No picture)

4. You will be taken through the signup process. Be sure to create a free Residential /
sipgate ONE account. Important: Make sure that sipgate assigns a phone number to you. It
is easy to get confused during the signup process and end up with an account that does not
have a phone number assigned. (No picture)

5. From the main page, select Settings in the upper right corner.

6. Take note of the phone number shown in the center of the screen. If you do not have a
phone number assigned, click "+ Phone Numbers" in the right navigation pane, make sure
that one is assigned, and take note of it. Next, click "SIP Credentials" in the right navigation
pane of the Settings page.

7. Review the SIP credentials shown in the pop-up window. Note that these credentials are
NOT the same ones you use to login to the sipgate website.
8. With your sipgate phone number noted and your SIP credentials (SIP-ID and SIP-
Password) also noted, you may log out of the sipgate website.

9. Visit sipsorcery.com and install the Microsoft Silverlight plug-in when prompted. Once
installed, create a new account. (No picture)

10. Once logged in, click on the SIP Providers menu.

11. There are two windows on the screen, an upper window and a lower window. Click Add
on the top row of the top window.
12. Create a new SIP Provider named "Sipgate" and enter the SIP-ID and SIP-Password into
the Username and Password fields in the pop-up window. As a reminder, you do NOT enter
the username and password you use to login to the sipgate website. The values you need
come from the SIP Credentials link on the Settings page of your sipgate account.

13. Click Update at the bottom of the pop-up window and watch the bottom window on the
screen, titled "SIP Provider Registrations". Within seconds of clicking Update, a new row
should appear in the window, titled Sipgate. Make sure that the checkbox in the Registered
column is checked. FREE CALLING WILL NOT WORK UNTIL THIS IS CHECKED.

14. Once the checkmark appears in Registered, you are ready to test your connection.
Download and install X-Lite from Counterpath. With X-Lite installed and running, right click
anywhere on the phone and choose SIP Account Settings. On the SIP Accounts screen, click
Add in the upper right corner (no picture).

15. Enter your SIP Sorcery account details as shown. Here you will use your SIP Sorcery
username and password from the website.

16. Click OK on the Properties of Account 1 screen and click Close on the SIP Accounts
screen. You will be returned to the telephone interface and should see a Ready status with
your SIP Sorcery username.
17. Use your cellphone to call the sipgate phone number you wrote down in step 6 above.
X-Lite should indicate an incoming call. Answer the call by clicking the green button. Make
sure you can hear sound in both directions.

18. Once you check the audio, return to the Google Voice website and add the sipgate line
to your Google Voice account. Click Settings in the upper right corner and then Phones in
the blue menu (if necessary). Click Add Another Phone at the bottom. Name the phone
Sipgate. Do not select Mobile as the type of phone. Google Voice will display a 2-digit
verification code and wait for you to click Connect. Click Connect and X-Lite will ring.
Answer the call and type in the 2-digit code displayed on the screen. The line will be added
to your list of managed phones.

19. With the line now verified in Google Voice, it is time to add dialout capabilities. Return
to section 10 of this document and copy the entire dialplan into your clipboard. Highlight the
entire script from "Begin Code" to "End Code" and hit Ctrl-C. Return to SIP Sorcery and click
Dial Plans.
20. The Dial Plans page will have window displayed. You will see one entry titled "default".

21. Click on that row and a pop up window will appear. Highlight the sample script shown in
the window, and hit Ctrl-V to paste the sample dial plan copied from this document. Return
to section 11 of this document and carefully follow the steps required to customize the
script. There are five pieces of information to enter: your home area code, your time zone,
your Google Account username, your Google Account password, and the 10-digit phone
number assigned to you by sipgate and tested in step 17. In the following two sections,
remove the "#" in blue highlight from the one line at the beginning of each line associated
with sipgate.

22. Once complete, click the Update button at the top center of the script editing window.
You should now be able to dial any US phone number directly from the keypad of X-Lite.
Test this by returning to X-Lite and dialing any known-good US-based phone number, such
as your cellphone.

14. Additional setup screenshots

This document is a collaborative effort, so we are constantly looking for other setup
instructions and screenshots. In particular, we would like to document the process for
configuring these two very popular clients:

• sipdroid for Android devices

• iSip for iPhone and iPod Touch
15. Revision History

12/04/2009 by norske - updates to the sipgate dialplan

• highlighted user-specific entries for the dialplan in blue (easier to know exactly what
should be changed to make the dialplan work for the new user)
• added some speeddial codes (including for 1-800-WEATHERCALL)
• replaced 747 (gizmo5) area code with 415 (sipgate) in what was line 80
• removed (2) # characters that were in front of the following two lines. They were
causing syntax errors in line 80 of the dialplan
when /(800|866|877|888|415)/ # US toll-free and sipgate-to-sipgate calls
route(5,"Destination - US toll-free or sipgate")
12/04/2009 (AM) by EasternPA -

• removed 911 from the sipgate and Gizmo5 dialplans until we have a better method
of hosting those life-critical services. Too risky otherwise IMHO.
• updated the formatting line for the speed dial sections of sipgate and gizmo5
dialplans
• replicated the highlighting in the Gizmo5 script
• copied the new speed dial entries from sipgate to gizmo5. I'm wondering if we
should build a separate section just for the speed dial directory to avoid having to
replicate the entries between the sections. Keep the unique entries in each dialplan,
but maintain a separate section for entries that are not unique to the SIP provider.
As more SIP providers get added to the doc, it will get messy trying to replicate a
long directory. That was also my original intention with creating the Wave. A
separate place for a common speed dial directory.
norske (6:30p): I agree that the speed dial section can remain fixed as-is. It
provides plenty of examples for others to tweak their own to suit their specific
needs. Another section would be OK - but especially useful if we have something
really useful that may otherwise be tough to find.
12/04/2009 (PM) by EasternPA -

• Renamed section 9 to be Gizmo5-specific

• Made extensive changes to section 10 to account for the recent spate of users
unable to complete the process

12/05/2009 by EasternPA -

• Removed all references to the 3-line dialplan. That short dialplan no longer works
(even with Gizmo5)
• Worked with norske to re-sequence Section 10. We agreed that there should be an
early test to ensure the path from sipgate to SIP Sorcery and the SIP client is up
and running. Then the sipgate line should be registered in Google Voice. Then the
dialplan should be copied over and customized.
• Replaced "your Google Voice number" with ".*" in sipgate dialplan per Red
Leatherman
12/08/2009 by norske (RE: the sipgate plan)
• Moved all user-required entries to the front-end of the dialplan - so user doesn't
have to make direct entries in the sys.GoogleVoiceCall line
• Adjusted the sys.GoogleVoiceCall line to accept named parameters from the initial
entry section

Added this bullet 12/10/2009:

• note the final two parameter on this line of code in the sipgate plan:

sys.GoogleVoiceCall(GV_USER,GV_PASS,SIP_NUMBER,@num,'.*',1,30)

The 1 designates the Phone Type, where 1 = Home, 2 = Mobile, 3 = Work, and 7 =
Gizmo. The default (no entry) is "2". For the sipgate plan, it could be that this
parameter is not an absolute requirement.

The need to add a Phone Type is a recent change.. and is documented here:

http://sipsorcery.codeplex.com/SourceControl/PatchList.aspx

With respect to the final parameter - it's the "callback timeout"in seconds which is the
amount of time the sipsorcery dialplan will wait for the GoogleVoice callback before giving up
and continuing with the next dialplan command. The timeout must be between 5 and 60s and if
not specified a default value of 30s is used."

I think that if you're OK with the default, then an entry isn't mandatory.

Here's a link to sipsorcery's dialplan help.. it's tough to read because of the gray
background, but it does explain key parameters used in the dialplans.

https://www.sipsorcery.com/help/dialplans.html

12/09/2009 by norske
• reposted the latest EasternPA document - that link was having issues this morning
• fixed a couple of formatting issues (things bold and larger font than they're
supposed to be)

12/10/2009 by EasternPA
• reformatted the new user credentials area at the top of the sipgate script (a few
lines were line-wrapped)
• updated the comments in the Google Voice section of the sipgate script since users
no longer need to enter their credentials in that section of the dialplan anymore

12/10/2009 by norske
revised the gizmo dialplan to match the change I made to the sipgate plan on 12/08/2009
as outlined above... namely:
• Moved all user-required entries to the front-end of the dialplan - so user doesn't
have to make direct entries in the sys.GoogleVoiceCall line
• Adjusted the sys.GoogleVoiceCall line to accept named parameters from the initial
entry section
12/11/2009 (AM) by EasternPA
Made MAJOR changes throughout the document. I created a new unified dialplan that is
portable to users of both Gizmo5 and sipgate. Every section from 7 on has been edited,
moved, or deleted. There are now two sets of two commented lines in the mandatory
editing section and one line in each set must be uncommented based upon the provider the
user has selected. I also nullified the checks for the AK, HI, Yukon, and toll free numbers
since they are all properly handled by GV. I left the original code in place as comments in
case anyone wants to experiment with it. One such change would be to specifically route
747 calls over Gizmo5, but only if a Gizmo5 account exists.

12/11/2009 (PM) by EasternPA

Made another round of major changes. I have incorporated most of norske's screenshots as
well as those provided by jamesgwvoice. I also resequenced the screenshots. For the users
who are able to successfully follow the sequence straight through and can make calls via X-
Lite by the end of section 11, they will see the screenshots for configuring the Linksys ATA.

For those users who aren't able to follow the steps in sections 8-11, they are referred to
section 13 for a more detailed set of instructions, including screenshots walking them
through the entire process.

12/12/2009 by norske
• Added links to the front section of Section 12:
- A wikipedia page on the PAP2T-NA
- Cicso support page, and
- User's guide to the Linksys PAP2T-NA
• Added some Linksys PAP2T-NA commentary at the end of Section 13

12/12/2009 by EasternPA
• Made a clickable Table of Contents