Set Up Your Own IP-PBX

Note: I know this is seriously out of date.  I have not been

able to update it, and I don't expect that to change for the
forseeable future.  Still, many people continue to find it
useful, so I will keep the page around for a while longer.   I
cannot provide help or answer questions on it, however.

The state of open source software has progressed to the point where you can set up your
own IP PBX at home in a single evening, with a minimum of investment needed.  All
you need is explicit instructions, which is what I provide here.

In past times, it would have been quite costly to have a “personal” PBX.  Using spare
hardware, open source software, and low-cost service providers, it can now be done for
almost nothing.  The key advance is the open source Asterisk IP PBX and the
Asterisk@Home package that includes Asterisk and a web-based GUI configuration
tool.

The Components
Your will have to set up three main components: the IP PBX itself, the phones (or
softphones) to be used with it, and the gateway service that lets you call other people on
the PSTN.  I will describe how to set up each of these.  I assume you already have a
home network and broadband access.  If you are behind a NAT firewall, it does not
matter – you don’t have to do anything special like running the IP PBX outside the
NAT.  All that matters is that you have enough bandwidth (upstream and downstream)
to carry voice traffic.

The IP PBX

You will need a computer to run the IP PBX.  While you may already have a Linux
server at home, I do not recommend using it to also run your phones.  If you want to do
that, you will have to figure out how to load and configure the PBX software on your
own, although the rest of the instructions here will be helpful.

If you have an old PIII class machine lying around, then use that.  If not, you can buy
one on EBay for about $40.

I used a Dell OptiPlex GX1, a PIII 450MHz system with 128M RAM.  It has a built-in
sound and Ethernet, so no additional hardware is required.
You will load Asterisk@Home on this computer.  It will take it over – it starts by
formatting the hard disk, so make sure there is nothing on the machine that you want to
keep.  Detailed instructions are given below.

Phones

You can buy SIP phones from Grandstream, SNOM, Cisco, even Avaya.  Or you can
buy an adapter from SNOM or Cisco and use an analog phone.  To get started, though,
it is easiest to get a softphone and run it on another computer.  We will describe how to
obtain and install the XLite softphone on a Windows machine to use for a phone.  You
will also use this Windows machine to administer the IP PBX, through a browser.  If
you want to use a Linux machine instead, you will need to get an appropriate softphone
for it, but you can still use its browser to administer the IP PBX.

SIP Gateway

I presume you want to communicate with others on the PSTN network, so you need to
obtain gateway service.  Since part of the call is being carried on the circuit switched
network, it costs real money.  This means you will have to pay for this part of the
system. 

I will describe how to set up access to two services, one for outgoing calls and another
for incoming calls.

VoipJet is used for outgoing calls.  It is priced at 1.3 cents per minute (USA), with no
monthly minimum or service charges.

BroadVoice is used for incoming calls.  You get a phone number (you can choose the
area code and exchange).  They have various plans, but I signed up for one that is $5.95
per month, with unlimited incoming minutes.  You can also make outgoing calls using
this account (it includes 100 minutes per month, and 3.9 cents per minute additional). 
There is a $9.95 one time activation fee to sign up for BroadVoice.

So if you have the hardware, you can set up your IP PBX for a total of $9.95 and run it
for $5.95 per month.    

Home Network

I assume you have a home network, and that you are setting up behind a Gateway
Router (otherwise known as a NAT firewall).   You will need to pick a static IP address
for your IP PBX that is on your home network. 

How Much Will This Cost?

I assume you have broadband service, a router, and a Windows machine to run the
softphone. 

If you already have a spare computer that you can dedicate to this project, there is no
cost at all for equipment, unless you need to buy an audio headset for the softphone.  If
you do not have a spare computer, then you can buy one on EBay for about $60,
including shipping.  This does not include a monitor, but you don’t need a monitor
except briefly when you first set it up.  I assume you can borrow a monitor (or that you
have a KVM switch).

Your only other initial cost will be the $9.95 activation fee to BroadVoice.

Your monthly cost will be $5.95 for incoming service from BroadVoice, and 1.3 cents
per minute for outgoing calls to VoipJet.  If you only make a few outgoing calls, you
could drop VoipJet, and stay within the 100 outgoing minutes that you get from
BroadVoice.  If, say, you use 100 minutes of VoipJet outgoing calls, it would cost an
additional $1.30.

So you can build and experiment with your own IP PBX for an investment of less than
$100, and for an ongoing cost of less than $100 per year.  This is a lot less than I am
now paying for my phone service from AT&T.  In fact, a whole year’s service would be
less than my monthly phone bill.  Hmmm…

Step 1 – Sign Up for Service

This section takes you through signing up with VoipJet and BroadVoice.  I am using
two service because

1. it is less costly, if you use it a lot, and

2. it illustrates how to set up two different kinds of trunks.

VoipJet

Browse to

http://www.voipjet.com.  Sine up for service. Then log on and follow the line describing
how to set up Asterisk. You will need to copy down your “VoipJet account number
(username),” your “Authorization code (password),” and your server IP address
(depending on your location).  Ignore the rest of the setup instructions.

BroadVoice

Browse to http://broadvoice.com.  Sign

up.  Say “I want to use my own SIP
device”.  When it asks what type of
device, select “Not Listed (Generic
SIP).  When it asks for details, just say
“Asterisk”.  Next, pick your phone
number.  Next pick your plan.  I recommend “BroadVoice BYOD Lite.”  Once you
have finished the sign up process, log in, click on “Account” and follow the “Show
Settings” link in the “Your Devices” section.  This section gives your phone number (in
case you forget) and your password (you will need this later). 

You should also click on the “Support Center” link on the left and bring up the “Step By
Step Installation Guide” for “Bring Your Own Devices.”  Select “Asterisk.”  You must
follow the instructions in Section 3 to find the “right proxy” to use.  This involves
pinging their proxies to find which has the lowest latency.  You will use this later.

            Detailed Instructions

1.      Ping the following hosts: proxy.lax.broadvoice.com,

proxy.dca.broadvoice.com, and
proxy.mia.broadvoice.com.

2.      Pick the one with the lowest latency.  In my case, it was
proxy.dca.broadvoice.com.

3.      Use nslookup to get the IP address of proxy.dca.broadvoice.com.  In

my case it is 147.135.0.128.

Step 2 – Set Up Home Network

Pick an IP address for the IP PBX.  You will need to find an unused address on your
subnet outside the range assigned by your DHCP server.  I picked 192.168.0.40.  If
you pick a different IP address, you will have to adjust the instructions accordingly.

Step 3 – Set Up SIP Softphone

Download and install XLite from http://xten.com. 

Run XLite.  Click on the “Menu” icon to configure

it.  Click on “System Settings”, then “SIP Proxy”, then “Default”. 

Fill in the following fields:

User name: 200

Authorization user: 200
Password: abc123
Domain/Realm: 192.168.0.40
SIP Proxy: 192.168.0.40

 
It should look something like this.

The phone will try to register, but for now it will fail.

Step 4 – Set Up Asterisk@Home

 

The rest of the instructions explain how

to install and configure
Asterisk@Home.  From here on, I will
refer to it as AAH. 

Introduction

AAH is a package consisting of several major components.  These were developed and
supported relatively independently.  The “Asterisk” part is the core IP PBX, and the
“@Home” part consists of applications, a provisioning system, an installer, and an
operating system that, together, make a complete package.  The major components that
make up AAH are

 Asterisk, the core PBX

 Sugar, a CRM system
 Flash Operator Panel, a screen-based operator’s console
  Web Meet Me Control, a meet me conferencing control application
 Asterisk Management Portal (AMP), a web-based provisioning tool for Asterisk
 A report system, part of AMP, which provides CDR reporting tools
 A Maintenance system, also part of AMP, which provides low level interfaces to
some components and real time system information
 CentOS, a version of Linux related to from Red Hat Enterprise Linux (but
without Red Hat branding and support).

Get the Software

AAH can be downloaded from http://asteriskathome.sourceforge.net.  Download the

basic ISO file, and burn it to a CD as a bootable volume (exercise left to reader).

Load AAH

This is going to reformat your hard disk and load everything from the operating system
on up, so make sure there is nothing on the hard drive that you want to save. 

Boot your machine from the CD.  When it prompts, type ENTER.  Then wait as
everything loads and compiles.  This could take 30 minutes or more, depending on how
fast your computer, hard drive, and CDROM are.  At the end, it ejects the CD and
reboots from the hard drive. 

The initial login is:

username: root

password: password

Set Up Networking

The machine probably got an IP address from DHCP, but it is not what you want.  Log
in as root and run:

netconfig

It will display setup information.  You should enter the following:

IP address: 192.168.0.40
 Netmask: 255.255.255.0

Gateway: 192.168.0.1

DNS Server: 192.168.0.1

OK these changes, then reboot the machine to make them take effect.

After reboot, long in once more.  You need to add the BroadVoice server IP address
to /etc/hosts.

Edit /etc/hosts, and add the following line at the end:

sip.broadvoice.com 147.135.0.128

Here the IP address was the one obtained when signing up with BroadVoice.

Once this is done, you can do the rest of the configuration through the web.

Browse To AMP

From your web browser, go to http://192.168.0.40.  You should see the following:
Select “Asterisk Management Portal.”  Log in as follows:

username: maint
password: password

Now you should be at the AMP main screen.

Click on “Setup” to bring you to the main setup screen.

 

Set Up BroadVoice Trunk

Click on “Trunks” and then “Add SIP Trunk.”  You see a blank SIP trunk form.
 You will need to fill out the main items on the SIP/Trunk screens. I am using
BroadVoice for incoming service only, so I will not administer outgoing trunk
information. 

Outbound caller ID:  7237570239

Max channels: 1

The outgoing settings can be left as is, except to fill in the trunk name as
BroadVoice.

In Incoming Settings, fill in the following:

User Context: 7327570239

begin_of_the_skype_highlighting              7327570239      end_of_the_skype_hi
ghlighting
User Details:
callerid=7327570239
context=from-pstn
dtmfmode=rfc2283
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
insecure=very
secret=********
type=user
user=7327570239
begin_of_the_skype_highlighting              732757023
9      end_of_the_skype_highlighting
username=7327570239
begin_of_the_skype_highlighting              732757023
9      end_of_the_skype_highlighting

The format of the Register String is:

username:password:phone_number@provider_domain.  For BroadVoice this is what
you will want to use (replacing with your BroadVoice phone number & password):

7327570239@sip.broadvoice.com:**********:7327570239@sip.broadvoice.com
/200

Once all of this information is entered, you can click the Submit button. Once you do,
a red line will appear at the top of the page. To "apply" your changes you must click
this red line (this will cause AAH to tell the Asterisk engine to reload its config files).

Set Up VoipJet Trunk

Click on “Trunks” and then “Add IAX2 Trunk.”  You see a blank IAX2 trunk form.

Fill in the fields as follows:

 Outbound Caller ID: 7327570239
Maximum channels: 1

The outgoing dial rules control how numbers are processed before they are sent to the
trunk.  In the case of VoipJet, phone numbers must be in the form “1+Area Code+Local
Number”.  If the number appears to have an area code but not a “1”, then it is added.  If
it does not have an area code, then “1732” is added (I am in area code 732).  So the dial
rules are:

1+NXXNXXXXXX
1732+NXXXXXX

This time we will fill in outgoing settings. 

Trunk name:: voipjet

Peer Details:
auth=md5
secret=***********  (get this from voipjet config info)
host=216.118.117.46  (get this from voipjet config info)
type=peer
username=***** (get this from voipjet config info)
notransfer=yes
context=default

You can leave the “Incoming Settings” and the “Register String” blank.

When you are done, the form appears as follows.

 
Set Up Extensions

Now it is time to define the extensions.  I will set up two extensions, number 200 in the
basement and number 201 in the study.  I have already shown how to configure XLite
extension 200, and you can configure another extension like it for 201.  Or you can set
up a SIP phone instead.

Click on “Extensions” on the left, and “Add an Extension.”  You will see the extensions
page.
 

The information you will add is the following:

Extension: 200
Password: abc123
Full name: basement
Voice Mail Password: 1234
Email Address: <your email address>

When you're finished adding the extension, you can see the details of the extension by
clicking on its name on the right-hand side of the AMP interface (as shown above).

Add extensions 201 similarly.  Remember to click on the red bar, to make Asterisk take
not of your changes.

Check for Phone Registration

At this point, the XLite application should have registered with AAH.  If not, exit XLite
(you have to stop it from the tray icon) and restart.  It should say “Logged in”.  If not,
see the troubleshooting section below.  To start with, make sure you have assigned the
same password in the extension form and in the phone.

On XLite, dial *23 to test audio input and output levels. 

Set Up Ring Group

Now we set up a ring group, so that we can ring all the extensions at once.  This is
useful, for instance, so that incoming calls can  alert at all extensions.

On the left, click “Ring Groups” and on the right “Add Ring Group”.
Fill out the form as follows:

Group Number: 1
Extension List:
  200
  201
Ring Time: 18
Destination if no answer: Voicemail basement <200>

>

Submit the changes, and click the red bar to make them take effect.

Set Up Digital Receptionist

A digital receptionist is an application that answers incoming calls, interacts with

callers, and forwards their call on.  It allows callers to dial an internal extension, to
access the company directory, or to reach selected destinations according to predefined
dial patterns.  We will set it up to dial by extension, to give the company directory, and
to ring all extensions if the caller presses “1”. 
The digital receptionist set up involves a series of screens and actions, not all of which
will be illustrated.  Part of the setup includes recording a message to be played to
callers.  You can record this independently, and download the MP3 file, or you can use
XLite to dial in and make the recording.  The latter method will be illustrated.  In
general, the setup instructions are clear and explicit, so this will only give the essentials.

Click “Digital Receptionist” on the left.  Enter 200 as the current extension number. 
On XLite, dial *77 and record a message.  It goes something like this:

Hello.  You have reached Charles Hayden’s experimental Asterisk phone system.  You
can dial a three digit extension.  Dial 1 to ring all extensions . Or dial pound for a
directory.

Hang up and dial *99 to hear the message.  Rerecord if necessary.

Name the recording “top greeting” and add a description.

On the next page, for “Number of options for Menu top greeting enter 1.

On the next screen, choose “Ring Group #1” as the action.  This means that when
the caller enters “1” it will go to Ring Group #1, which rings all extensions.

Set Up Outbound Routing

Click on the “Outbound Routing” link and fill in the following:

Route Name: outgoingVoipJet

Dial Patterns:  011.
                1NXXNXXXXXX
                732NXXXXXX
Trunk Sequence: IAX2/voipjet

The dial patterns given here will allow you to dial international calls, long distance with
1+area code, and calls within the 732 area code with only the area code but without the
1.  This is how it works here in the 732 area.  If you can dial locally without the area
code, then you could add “NXXXXXX” as well.   As we have seen, the trunks cannot
necessarily accept all these dialing combinations, so they have their own rules to screen
and transform them.
If you've made it this far you should be able to dial "7777" on your SIP phone - this will
simulate an incoming phone call - and you should hear your greeting. At this point the
system should have enough functionality to dial out through VoipJet.

Go ahead and call a number and it should ring at the other end. Remember that you need
to dial 1 followed by a 10 digit number. If it does then pat yourself on the back! Only a
few more steps are needed to get incoming calls to work (you can try to call your
BroadVoice number, but it won't work yet).

Set Up Incoming Calls

Now we will set up incoming calls to go to the digital receptionist.  Choose the
“Incoming Calls” link, and click the “Digital Receptionist” “top greeting”.
Edit extensions.conf

So far we've done everything through the AMP web GUI. Now we have to drop into
editing text files. This is obviously not desirable, but this is how it is right now, so you
have to go through this.  You can edit extensions.conf through the AMP web GUI. 
Click the Maintenance link at the top of the AMP page and then click the Config
Edit link in the left-hand menu. Then click on extensions.conf.

Look on the left for from-sip-external and click it.  By default, AAH sends all
incoming SIP calls to a congestion status. This means that all incoming calls will ring
fast-busy to the caller and they will never make it into our AAH system.

There are four lines under the [from-sip-external] section. One is already
commented out with a semi-colon (;). Comment out the remaining three lines by adding
semi-colons. Now we need to add two new "exten" lines.

The first is:

exten => _.,1,Wait(1)

 
which tells AAH to wait for 1 second once a call has been detected from an external SIP
device. This makes sure we don't clip off the first part of our greeting message. The
second line we need to add is:

exten => _.,2,Goto(from-pstn,s,1).

When we're done the section should look like this:

[from-sip-external]
;give external sip users congestion and hangup
;exten => _.,1,AbsoluteTimeout(15)
;exten => _.,2,Congestion
;exten => _.,3,Hangup

exten => _.,1,Wait(1)

exten => _.,2,Goto(from-pstn,s,1)

When you're finished and you've saved your changes, you need to click the link at the
top of the config edit screen labled Re-Read Configs. This will cause Asterisk to
reload its configuration files and make any changes take effect.

Test Incoming Calls

Now is the moment of truth. If everything went smoothly, you should now be ready to
test an inbound call to your AAH system. From a phone (POTS or cell, either will do)
dial your BroadVoice number. You should hear your greeting! At this point you'll
probably be giddy that it's all working. Go ahead and listen to your greeting a few times
(dial 9 to repeat it). Then either dial "1" or your extension and your SIP phone should
start ringing. You now have Asterisk@Home configured and working with your
BroadVoice account.

Set up Passwords

You need to change the default passwords, so that people do not break into your
system.  There are a variety of passwords, used for different things.

AMP allows you to assign different login accounts to access the Maintenance section
separately from the other sections (Setup, Reports, and Panel).  If you log into AMP
with the maintenance account, you will be able to access everything, but if you log in
with the admin account, you will have to log in also to get to the maintenance section.

Username Default Command to Change Usage

Password Password
root password passwd log in to console
wwwadmi password password-amp log into AMP GUI,
n access Setup,
Reports, Panel
maint password password-maint log into AMP
 Maintenance section
meetme ?? passwd-meetme log into Web Meetme
Control
admin password passwd admin for checking system
mail (Maintenance /
Web Mail login)

To change these passwords, you must log into the console as root, either on the physical
console or using a ssh client such as putty (which can be obtained from
http://www.chiark.greenend.org.uk/~sgtatham/putty/) from a windows platform or using
ssh from a Linux platform.

Troubleshooting
If things did not work the first time, you can track down the trouble using the Asterisk
console.  This can be used to display SIP messages coming through the system, as well
as steps executed in interpreting the dial plan instructions. 

If you are more comfortable using Ethereal, you can use that instead (or also) to see the
SIP messages.  I will not describe how to download or configure it, other than to point
out that it can be obtained from http://www.ethereal.com.

To use the Asterisk console, go to the Linux console or enter through ssh.  Remember
the username is root and the password is password.  Give the command:

asterisk –vvvvr

This will attach to the console with verbose mode set.  Give the command

sip debug

to enable SIP debugging.  You can turn off SIP debugging using the command

sip no debug

Other console commands are summarized at http://www.voip-info.org/wiki-

Asterisk+CLI.

Softphone will not register

XLite registers by sending a REGISTER message, which is challenged.  It then sends

another REGISTER with credentials, which is accepted.  If you do not see these
REGISTER messages, then XLite proxy setting is probably not right.  If you see the
challenge but it still rejects the REGISTER with credentials, the passwords are probably
mismatched.  The password abc123 should be used in both places. Also, be sure the
phone “Authorization user” is set to 200.
Outgoing calls

If you are having trouble making outgoing calls, check the username and secret in the
IAX2 trunk.  Make sure you can ping the host address listed there.  The username is a 4
digit number that you find from your account settings on VoipJet. 

On a message trace, you should see an INVITE going out, followed by receipt of
TRYING and OK (when the far end answers).  If there is no response, then perhaps you
have the wrong IP address.  If the INVITE is immediately followed by a 4XX or 5XX
error, maybe the username or password is wrong.

Incoming calls

If the incoming BroadVoice trunk is set up properly, Asterisk will register successfully,
and will renew the registration every 10 seconds.  Look for an outgoing REGISTER, a
challenge, another outgoing REGISTER with credentials, and an incoming 200 OK.  If
you get no response, or either REGISTER yields an error reply, then look at the host,
secret, and user fields from the SIP trunk. 

If the REGISTER succeeds, then when you call (from a POTS phone) you should see an
INVITE being received.  If you have not properly changed extensions.conf, then
the system will reply with an error rather than with TRYING.  Look at the debug output
to see how the INVITE is being processed.  You should see the Wait and the GoTo
that you entered into extensions.conf.  If you see these, and the call is still not
answered, you might want to change Incoming Calls to route the call to a specific
phone, for instance to “basement <200>”.  This will bypass the Digital Receptionist
and the Ring Group for now.  If this fixes things, look at those two screens. If not, make
sure the “context=from-pstn” is present in the incoming trunk user details.  If this
still does not fix things, you will have to follow the trace as it steps through the contexts
and steps in extensions.conf.

More

This section describes some things you might want to do to further customize your
system.  I am experimenting myself, and presenting the results here.  These instructions
might not be the most effective way to achieve the desired results.  If you find a better
way, please let me know.  The instructions in this section are briefer, reflecting your
increased level of experience with AAH.

Weather for external callers

You can modify the digital attendant so that it can read the weather to callers. 
Remember to modify your digital attendant announcement recording if desired (“press
2 for weather”).

1. Modify the Digital Receptionist.  For “Number of options” give “2”.  On the
next screen, for option 2, give “Custom App” and enter

custom weather,s,1
 2. Go into “Maintenance”, “Config Edit”, “extensions_custom.conf”.  At the end,
add:

[custom-weather]
exten => s,1, Goto(from-internal-custom,*61,1)

You can customize the weather for your own city be editing
/var/lib/asterisk/agi-bin/weather.agi.  You need to edit $custpath
and $filename.  Don’t forget to modify the announcement a few lines down.  You
can ftp to weather.noaa.gov and cd to data/forecasts/city to see the
state/city combinations that are available.

Outgoing BroadVoice

You can set up the BroadVoice trunk to carry outgoing calls.  Since you get 100 “free”
minutes per month, you might want to use these up first, before switching over to
VoipJet.  It would be great if the system could do this for you, but for now there is no
easy way to do this.  So instead, let’s up a dial-9 trunk for BroadVoice.

1. Open the BroadVoice trunk.  Add “Peer Settings” of:

allow=ulaw
authuser=7327570239
callerid=7327570239
disallow=all
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=7327570239
host=sip.broadvoice.com
insecure=very
qualify=yes
secret=*******
type=peer
username=7327570239

2. Go to “Outbound Routing”.  Add a new route.  Name it “outgoingBroadVoice.” 

Add the following dial patterns.  These mimic the local dialing rules, which
permit “1” followed by a ten digit number, or “732” followed by a seven digit
number.  An initial “9“ selects the broadvoice trunk, but it is stripped off before
the number is dialed.

9|1NXXNXXXXXX
9|732NXXXXXX
  

Set the “Trunk sequence” to SIP/broadvoice.”

Set Up Voice Mail

By default, when you set up an extension, and add an email address in the “email
address” field, then when the system records a message, it sends email containing a link
that will let the recipient listen to the voice message on their computer (through the
browser).  For this to work, you must open up your Asterisk machine to browser traffic
from outside the router’s firewall.  I leave it to the reader to figure out this one.

Once you do, you still need to change the email, which contains a link to the mail
reading web page. 

1. Go to “Maintenance” and then “Config Edit” and choose “vm_email.inc”. 

2. Replace the string “192.168.1.101” with the external name or IP address of your
Asterisk system.  In other words, you need to form a URL that can be used to
browse to the Asterisk system from where you will be reading the mail, which is
probably outside your NAT router.
3. After you click “Update” do not forget to select “Re-Read Configs”.
4. Test it by calling extension 200, leaving voice mail, receiving the email, and
following the link.
5. You will be prompted to log in.  Use extension number 200, and its password
“1234”.  You can then retrieve and listen to the email.

You may notice a few things wrong with the voice mail reader application.  I guess this
means that it is not really finished, although it appears to basically work.  I noted that
the initial login does not fill in your extension, as it should.  Also the image links are
broken.  Maybe the next version will clean up these details.

Add Free World Dialup IAX Trunk

Free World Dialup (FWD, at

http://www.freeworlddialup.com/
) is a free VoIP service provider. 
Using it, you can call other FWD
users, and through cooperative
arrangements with other service providers, subscribers on many other VoIP networks. 
In addition, you can use their gateways to call to the PSTN (but only toll free numbers,
since someone still has to pay for other kinds of calls).

You first need to sign up for a FWD “phone number,” which you can do at the web site
listed above.  Be sure to enable IAX service while you are there.  My number is
679263.  I will use this in the examples below, but you should replace it with your own
number and password, as appropriate. 
1. Open the Trunks page, select “Add Trunk” and select “Add IAX2 Trunk”.  Enter
the following information.  I have chosen to use a prefix of “393” (FWD) to
select the FWD trunk.  You can pick a different trunk prefix if you want. 

Outbound Caller ID: Charles Hayden <679263>

Maximum Chanels: 1

Trunk Name: fwd

PEER Details:

allow=ulaw

auth=md5

disallow=all

host=iax2.fwdnet.net

qualify=yes

secret=<your FWD password>

type=peer

username=679263

USER Context: iaxfwd

USER Details:

allow=ulaw

auth=rsa

context=from-pstn

disallow=all

inkeys=freeworlddialup

secret=<your FWD password>

type=user

Register String:

679263:<your FWD password>@iax2.fwdnet.net

2. Open the “DID Routes” page, and select “Ad Route”.  The following
information:

DID Number: 679263

Ring Group: #1

3. Open the “Outbound Routing” page and select “Add Route”.  Enter the
following information:

Route Name: outgoingFWD

Dial Patterns: 393|.

Trunk Sequence: IAX2/fwd

After you submit and apply these changes, you can go to the “Maintenance/Asterisk
Info” page and check the IAX2 Sip Registry and the IAX2 Peers.  These should both
show the trunk you set up with State Registered and Status OK.  You can test FWD
by dialing “393612” to hear the time.  You can make an incoming call to your Asterisk
system using the FWD page that used to sign up for IAX service.

ArtDio

There are many choices for phones that you can use with the Asterisk system.  Among
the possibilities are IAX phones, SIP phones, SIP softphones, and analog phones with a
SIP/analog converter.  Such a converter should provide a FXO interface.  Sources of
phones and converters are listed below.

I purchased some ArtDio phones, since they were the least

expensive SIP phones I could find.  Adding new extensions
for the phones is straightforward – just like the others. 
Administering the phone itself was a little more difficult.
 

The procedure for setting up the phone consisted of two steps.

1. Plug the phone in, log on to it, and set its IP address, net mask, and default
gateway parameters.
2. Browse to the phone and set up its SIP gateway, its name and password, and
other SIP parameters.  You will know these are right when the phone can
register.  In the case of the ArtDio phone the critical step was knowing that there
were two passwords, and that only the “super user” password (“12345678
begin_of_the_skype_highlighting              12345678      end_of_the_skype_highl
ighting”) would allow access to the key SIP configuration parameters. 
3. The fields that I had to change on the ArtDio are as follows (assuming the
extension is 201 and the IP address of the phone is 192.168.0.60):

Use service: checked

Service type: common
Service addr: 192.168.0.40

Service id: 192.168.0.40

Phone number: 201
Account: 201
PIN: abc123
Call type: advanced

If you do not see the screen below when you browse to the phone’s IP address, but
instead see a simpler screen, then it means that you have logged in with the phones user
password, not its superuser password.

 
 

Sipura

Once I had a few phones set up, I realized that there are
places in my home where I wanted a phone, but where I did
not have Ethernet wiring.  There are several ways to address
this:

 rewire my house, putting Ethernet jacks all over

 use an analog phone adapter and set up cordless phones as extensions
 get WiFi SIP phones, used in conjunction with my existing wireless
infrastructure
 use an analog phone adapter and existing phone wiring and phones

The simplest, least expensive, and least disruptive choice was to use an analog phone
adapter with a cordless hone.  The Sipura adapter is widely available on eBay.  It was
distributed by various IP telephony service providers. I guess people who tried it and
did not like it are selling their used adapters.  Just make sure you get one that is
“unlocked”.  Apparently Vonage adapters are set up so that you cannot modify the
critical SIP parameters, and are not usable with Asterisk.  Other service providers such
as BroadVoice, do not lock their devices.

I bought a used Sipura SPA-2002.  I have also tried this with a Sipura 1000 and it works
about the same.  Setting it up was pretty easy. 
1.      Set up an extension, as before, in the AMP Setup/Extensions screen. 

2.      Plug it into Ethernet and connect a phone.  When you apply power, it gets an
IP address from DHCP. 

3.      You need to get the IP address.  Pick up the phone, dial “****” and then dial
“110#”. It will speak the IP address.

4.       Browse to the address as follows:

http://<ip_addres>/admin/advanced

5.      If you have purchased a used unit, it would be a good idea to clear out any old
settings first.  On Sipura devices, you can do a factory reset by:

a.       Dial “****” and verify that you hear a voice prompt.

b.      Dial “RESET#” (73738#).

c.       Dial “1” to confirm.

6.      On the “System” tab, enter the fixed IP address you want to assign the device.

When you are done, submit the changes.

7.      Using the new IP address, browse to the advanced admin screen again.  This
time, select the “Line1” tab.  Enter the proxy information (the IP address of the
Asterisk system) and the subscriber information (the extension number and
password you set up in step 1).
 You should be able to use the phone at this point.  Go to AMP’s
Maintenance/Asterisk Info screen to check that the new extension is registered.

Router

Once you have your system set up and are using it for
phone service, you may notice that the sound quality is
terrible when you are doing other things on your broadband
connection, such as large FTP transfers.  This depends a lot
on your connection speed.  In my case, which I imagine is
pretty common these days, I have only 128K bits per
second uplink speed.  For comparison, with the standard
G711 coder, one phone call takes 64K bits, half the
bandwidth.  Unless you do something, FTP will take
enough bandwidth to make telephone calls impossible. 
What can you do about this ?  Your router may be able to
help, by giving traffic coming to and from the Asterisk box priority. I have a Linksys
WRT54G, a widely available combo router, 4 port switch, and wireless access point.  It
has a way to assign priority to different traffic in various ways.  I set it up to give
priority to a given switch port, where I have the Asterisk system plugged in. 

To set this up, browse to your router, log in, and go to “Applications and Gaming” and
then to “QoS”.  I plugged my Asterisk system into port two, so I set up:

QoS: Enable
Port 2: Priority High
 

Next
The AAH people have provided you with a ton of pre-configured features. You can start
reading through the main configuration files to see some of the cool things you can do
with AAH and your BroadVoice and VoipJet accounts. You might want to set up the
BroadVoice trunk to do outgoing calls, using a prefix.  It would be nice if it could use
BroadVoice for outgoing calls until the 100 minutes are used up, then switch to
VoipJet.  If you figure out how to do this, let me know.

You can make a backup of your configuration files by backing up the files in the
/etc/asterisk directory of your AAH server. The AMP “Backup” command on
the left can be used to set up periodic backups.

If you are interested in making international calls, add some more patterns in
“Outbound Routing”.  Commonly used dial patterns are preconfigured, so you
only have to select them.

Read through the extensions.conf and see what all is already configured for you
to test and work with.

Further Reference
Here are some places to go to find out more about AAH, Asterisk, and the components
that make it up.

Service Providers
 BroadVoice http://broadvoice.com
VoipJet http://voipjet.com
VoicePulse http://connect.voicepulse.com
TelaSip http://www.telasip.com/index.html?PARTNER=coupon4voip
Free World Dialup http://www.freeworlddialup.com
LibreTel http://www.libretel.com

Software

Asterisk@Home and Asterisk:

Asterisk@Home http://asteriskathome.sourceforge.net/
Asterisk http://www.asterisk.org/
Asterisk@Home Handbook http://asteriskathome.sourceforge.net/handbook
Asterisk@Home support http://www.voip-info.org/tiki-index.php?
page=Asterisk+at++Home
Asterisk Support http://www.voip-info.org/tiki-index.php?page=Asterisk
Asterisk@Home info http://geekgazette.com/index.php?
option=com_frontpage&Itemid=1&limit=4&limitstart=8

AMP (Asterisk Management Portal)

AMP Home http://amp.coalescentsystems.ca/

FOP (Flash Operator Panel)

FOP Home http://www.asternic.org/

Sugar CRM

What it is and how to set it up http://www.sugarcrm.com/home

CentOS

            A free enterprise OS http://centos.org

Phones

XLite http://xten.com/index.php?menu=products&smenu=xlite
SipXPhone http://www.sipfoundry.org/sipXphone/index.html

Tools

            Ethereal http://www.ethereal.com

            Putty http://www.chiark.greenend.org.uk/~sgtatham/putty/

Phones

Buy phones http://www.voiplink.com/

Buy phones http://www.voipsupply.com/
  Buy phones https://www.thevoipconnection.com/
Buy Phones http://store.sipphonestore.com/
XLite softphone http://www.xten.com/
SNOM phones http://www.snom.com/phones.html?&L=1
ArtDio phones http://www.artdio.com.tw/en/ipf.htm
Grandstream phones http://www.grandstream.com/y-product.htm
Cisco phones http://www.cisco.com/en/US/products/hw/phones/ps379/
Avaya phones http://www.avaya.com/gcm/master-usa/en-
us/products/offers/4600_series_ip_telephones.htm&View=ProdTechSpec

Acknowledgements
A preliminary version of this set of instructions was produced by Jacob Cazell
(http://www.cazz.org/).  It helped immensely in getting my own system set up properly
the first time, and inspired this document.  A few of the sections here incorporate
material that first appeared there.

I will be adding to these instructions from time to time as I experiment with additional
features, service providers, and subsequent releases.  If you have suggestions,
corrections, or comments on these instructions, I welcome your feedback.  Please send
commands to cchayden@gmail.com

Appendix 1 – Feature Codes

*411 Directory
*43 Echo Test
*60 Time
*61 Weather
*62 Schedule wakeup call
*65 Festival test (your extension is XXX)
*69 Last Called ID
*70 Activate Call Waiting (deactivated by default)
*71 Deactivate Call Waiting
*72 Call Forwarding System
*73 Disable Call Forwarding
*77 IVR Recording
*78 Enable Do-Not-Disturb
*79 Disable Do-Not-Disturb
*8 Pickup
*90 Call Forward on Busy
*91 Disable Call Forward on Busy
*97 Message Center (does no ask for extension)
*98 Enter Message Center
*99 Playback IVR Recording
666 Test Fax
700 Park
7777 Simulate incoming call
888 Barge In
  

Appendix 2 – Asterisk CLI commands

This is a selected summary of the Asterisk CLI commands.  These commands can be
entered on the Asterisk console, which you access by running:

asterisk –r

General commands
!<command>: Executes a given shell command

abort halt: Cancel a running halt

add extension: Add new extension into context

add ignorepat: Add new ignore pattern

add indication: Add the given indication to the country

debug channel: Enable debugging on a channel

dont include: Remove a specified include from context

help: Display help list, or specific help on a command

include context: Include context in other context

load: Load a dynamic module by name

logger reload: Reopen log files. Use after rotating the log files.

no debug channel: Disable debugging on a channel

pri debug span: Enables PRI debugging on a span

pri intense debug span: Enables REALLY INTENSE PRI debugging

pri no debug span: Disables PRI debugging on a span

remove extension: Remove a specified extension

remove ignorepat: Remove ignore pattern from context

remove indication: Remove the given indication from the country

save dialplan: Overwrites your current extensions.conf file with an exported

version based on the current state of the dialplan. A backup copy of your old
extensions.conf is not saved. The initial values of global variables defined in the
[globals] category retain their previous initial values; the current values of global
variables are not written into the new extensions.conf. Using "save dialplan" will
result in losing any comments in your current extensions.conf.

set verbose: Set level of verboseness

show agents: Show status of agents

show applications: Shows registered applications

show application: Describe a specific application

show channel: Display information on a specific channel

show channels: Display information on channels

show codecs: Display information on codecs

show conferences: Show status of conferences

show dialplan: Show dialplan

show image formats: Displays image formats

show indications: Show a list of all country/indications

show locals: Show status of local channels

show manager command: Show manager commands

show manager connect: Show connected manager users

show parkedcalls: Lists parked calls

show queues: Show status of queues

show switches: Show alternative switches

show translation: Display translation matrix

soft hangup: Request a hangup on a given channel

show voicemail users: List defined voicemail boxes

show voicemail zones: List zone message formats

Server management
restart gracefully: Restart Asterisk gracefully

restart now: Restart Asterisk immediately

restart when convenient: Restart Asterisk at empty call volume

reload: Reload configuration

stop gracefully: Gracefully shut down Asterisk

stop now: Shut down Asterisk immediately

stop when convenient: Shut down Asterisk at empty call volume

extensions reload?: Reload extensions and only extensions

unload: Unload a dynamic module by name

show modules: List modules and info about them

show uptime: Show uptime information

show version: Display Asterisk version info

AGI commands
show agi: Show AGI commands or specific help

dump agihtml: Dumps a list of agi command in html format

Database handling commands

database del: Removes database key/value

database deltree: Removes database keytree/values

database get: Gets database value

database put: Adds/updates database value

database show: Shows database contents

IAX Channel commands

iax2 debug: Enable IAX debugging
 iax2 no debug: Disable IAX debugging

iax2 set jitter: Sets IAX jitter buffer

iax2 show cache: Display IAX cached dialplan

iax2 show channels: Show active IAX channels

iax2 show peers: Show defined IAX peers

iax2 show registry: Show IAX registration status

iax2 show stats: Display IAX statistics

iax2 show users: Show defined IAX users

iax2 trunk debug: Request IAX trunk debug

iax debug: Enable IAX debugging

iax no debug: Disable IAX debugging

iax set jitter: Sets IAX jitter buffer

iax show cache: Display IAX cached dialplan

iax show channels: Show active IAX channels

iax show peers: Show defined IAX peers

iax show registry: Show IAX registration status

iax show stats: Display IAX statistics

iax show users: Show defined IAX users

init keys: Initialize RSA key passcodes

show keys: Displays RSA key information

SIP channel commands

sip debug: Enable SIP debugging

sip no debug: Disable SIP debugging

sip reload: Reload sip.conf  

sip show channels: Show active SIP channels

 sip show channel: Show detailed SIP channel info

sip show inuse: List all inuse/limit

sip show peers: Show defined SIP peers (clients that register to your Asterisk
server)

sip show registry: Show SIP registration status (when Asterisk registers as a client
to a SIP Proxy)

sip show users: Show defined SIP users