3  Installation
         (The dumb-me way)

Download the ISO from here

http://www.trixbox.org/modules/smartsection/item.php?itemid=2/

 

 

 

 

3.1  Change default Settings

Once TRIXBOX has been installed, some default system changes need to be made to TRIXBOX.

Log in to your new TRIXBOX box (user: root, password: password)  

      

Before you proceed any further, update TRIXBOX – go to section UPGRADING TRIXBOX - it will make you a happier digger :)

3.1.1  To get Help

At the command line, type  

help-trixbox  

A list of help will be displayed – see illustration below;  

 

3.1.2  Change IP Address (set IP address to Static)

Change Asterisk IP address from DHCP to Static.  At the command prompt enter:

Netconfig

          

Select [Yes] to set up networking and hit enter.

You will then see the following screen.

           

Use the Tab key to cycle through the fields. Enter the IP address that is to be allocated to the Asterisk box, the Netmask (subnet mask), Default Gateway and Primary nameserve as per the example above.  In my example above, I used my existing network regime.

Once done, select OK.

Those are the initial inputs that require immediate attention.  Others, such as Admin password etc, are optional although it is recommended that you change them.  Once done, log off Linux and reboot.

Shutdown –r now

Asterisk will now start with the new IP address.

 

3.2  Configure TRIXBOX using Web Browser

Now you can connect to http://ipaddress/  (e.g. http:192.168.1.100) to configure TRIXBOX.  You will be presented with the Main Menu screen as illustrated below.

   

                                                              Main Welcome Screen

 

3.2.1  Log in to System Administration

**To log in to System Administration, use user: maint, password: password unless you have changed the password during initial set up in 3.1.2  

Once you logged in, you will be presented with the following screen,  

Main Configuration Screen

 

3.2.2  Configuring Trixbox using freePBX

At this stage select the freePBX option and you will be presented with the screen.

From now on, we will be configuring Trixbox using freePBX web GUI.

You are now in freePBX and this is where most of your configurations will start.

Select Tools  

The red bar will appear.. click on it to begin

Click Tools again

The same screen will appear without the red bar on top.  

You will need to activate all the Administration/Management modules that you require eg: Extensions, Queue, Digital Receptionists, Trunks etc.  

To do this, select Module Admin option on the left right under the FreePBX decal and you will be presented with the Module Administration screen.  You will be required to enable and installs all the modules that you are likely to need.  It does not matter if you enabled modules that you do not require as all it will do is making FreePBX refresh a little slower.  

Click on Module Admin option on the left.

You will notice all the modules that are available as illustrated below.

Note:  You don’t have to select all the modules.  Only select the modules that you need.  

To make it simple, I selected all modules by ticking all the tick boxes.

Click on the Submit button

However you may, at a later stage, choose to disable some of the modules that you will not need by ticking the box next to the module and from the dropdown box next to the submit button, select “Disable Selected” and hit Submit.  

Connect to online module repository to update any of the modules that have been updated since the installation, or include any other extra modules that have been released.

Click on the link pointed by the red arrow in the illustration above.

When connected, you will be presented with the following screen:

Apart from the modules that are already enabled, you will also see more modules that you can install and also modules that have been updated (highlighted in amber) since your installation.

Again to simplify matter, I will select all the modules.

First I will select all the Yellow modules and hit the submit button.  When the screen refresh, I will select all the other modules beneath it and hit the submit button.

I now have all modules installed and enabled.

Click on the red bar to commit the updates.  When the screen refreshed, you are done with the modules management section.

You will now notice more options have appeared on the left side of the screen.

Once you have enabled all the required modules, you are ready to start.  From this point onward, the set-up process are similar the difference being the presentation of the management screen and some extra options in FreePBX.

Now you can start configuring TRIXBOX.  Notice the selection options on the left.  Selecting each option will display configuration screen for that particular function e.g. creating new extensions, creating new trunks etc.

This is where most of the action begins.

3.3  General Settings

The first thing I do is select General Setting and set it up as illustrated below.

It is self-explanatory so I will not try to explain, as it is quite minimal and nothing substantial that warrants explanation.

Notice that the extra information that will define the way Asterisk behave, are also required in freePBX.  Set the fields to the following (these are vital information):

 

Hovering your mouse on the corresponding field description with a yellow/amber underline will display the purpose of the fields.

 

Asterisk Outbound Dial command option: “r” which generate the ring when you dial out, or “m” if you want music instead.  There are other options of course – refer to the chapter on Dial Command Options.

Country Indications: Australia

Allow Anonymous Inbound SIP Calls?: Yes (if this is not set you ‘Yes’, all inbound unidentified SIP calls will not be accepted

After setting up the General Settings, click on Submit Changes button and the red bar on top of the screen for the change to take effect.

    3.3.1  Dial Command Options

In the Asterisk Dial command option, you may customise your preference to the way asterisk behave e.g. if you want the caller to hear music instead of the standard ringing sound, you may replace the “r” with an “m”.  For further options, hover your mouse on the label and you will be informed of the other options.

The following are the dial command options available to you:

Options:

A(x)

Play an announcement to the called party, using 'x' as the file

C

Reset the CDR for this call

d

Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists.

D([called][:calling])

Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone.

f

Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller.

g

Proceed with dialplan execution at the current extension if the destination channel hangs up

G(context^exten^pri)

If the call is answered, transfer both parties to the specified priority. Optionally, an extension, or extension and context may be specified. Otherwise, the current extension is used.

h

Allow the called party to hang up by sending the '*' DTMF digit.

H

Allow the calling party to hang up by hitting the '*' DTMF digit.

i 

Jump to priority n+101 if all of the requested channels were busy.

L(x[:y][:z])

Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option:

  • LIMIT_PLAYAUDIO_CALLER yes|no (default yes) - Play sounds to the caller.

  • LIMIT_PLAYAUDIO_CALLEE  yes|no - Play sounds to the callee.

  • LIMIT_TIMEOUT_FILE   File to play when time is up.

  • LIMIT_CONNECT_FILE   File to play when call begins.

  • LIMIT_WARNING_FILE   File to play as warning if 'y' is defined.  The default is to say the time remaining.

m([class])

Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified.

M(x[^arg])

Execute the Macro for the *called* channel before

connecting to the calling channel. Arguments can be. specified to the Macro using '^' as a delimeter. The Macro can set the variable MACRO_RESULT to specify the following actions after the Macro is finished executing

  • ABORT    Hangup both legs of the call.

  • CONGESTION   Behave as if line congestion was encountered.

  • BUSY   Behave as if a busy signal was encountered. This will also have the application jump to priority n+101 if the 'j' option is set.

  • CONTINUE  Hangup the called party and allow the calling party to continue dialplan execution at the next priority.  

  • GOTO:<context>^<exten>^<priority> - Transfer the call to the specified priority. Optionally, an extension, or extension and priority can be specified.

n 

This option is a modifier for the screen/privacy mode.
It specifies that no introductions are to be saved in the
priv-callerintros directory.

N

This option is a modifier for the screen/privacy mode.  It specifies that if callerID is present, do not screen the call.

o 

Specify that the CallerID that was present on the
*calling* channel be set as the CallerID on the *called*
channel. This was the behavior of Asterisk 1.0 and
earlier.

p 

This option enables screening mode. This is basically
Privacy mode without memory.

P([x]) 

Enable privacy mode. Use 'x' as the family/key in the
database if it is provided. The current extension is used
if a database family/key is not specified.

r 

Indicate ringing to the calling party. Pass no audio to
the calling party until the called channel has answered.

S(x) 

Hang up the call after 'x' seconds *after* the called
party has answered the call.

t

Allow the called party to transfer the calling party by sending the DTMF sequence defined in features.conf.

T

Allow the calling party to transfer the called party by sending the DTMF sequence defined in features.conf.

w Allow the called party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf (default *1 in Asterisk v1.2).

W

Allow the calling party to enable recording of the call by sending the DTMF sequence defined for one-touch recording in features.conf (default *1 in Asterisk v1.2).

    

3.4  Extensions

The number of extensions to be set up depends on you. You can have soft phones installed in 4 or 5 computers or mixture of  ATAs and SIP SoftPhones.  In my case I have 4 extensions to experiment with – 3 soft phones and one ATA.

There are a number of extension numbers you should avoid using unless you are prepared to edit and change some codes.

 

3.4.1  Extension numbers to avoid

It’s best to avoid the following extension numbers:

200                  -           Park Notify
300-399            -           Reserved for speed dial
666                  -           Reserved for FAX testing
70-79               -           Reserved for calls on hold
700-799            -           Reserved for calls on hold
7777                -           Reserved extension for incoming calls simulation

3.4.2  Create Extensions

To create extensions, select the type of trunk e.g. SIP, IAX2, ZAP or Custom, is done from the Create Extension main menu illustrated below:

 

The illustration below is where you create the extension.

Submit when done.

 

                                   Add Extension Screen

Click on the red bar on the top of the screen every time you create a new extension.

 

Click on the Add Extension button to add more extensions.

My extensions are 2000, 2001, 2002 and 2003

For simplicity, I allocated passwords to be the same as the extension numbers. If you enabled Voicemail, you may allocate the same password as well but you don’t have to. You may also nominate an email address for Voicemail Email Notification – it’s up to you.  This is covered in more detail in the Chapter a little later in this document.


Note:
If you want this extension to be a remote extension, you will need to edit the extension and change the entry under Device options.  Nat:  Never to Nat:  yes.
 

3.5  Follow Me

After setting up your extensions, you need to decide if you want Asterisk to call another pre-arranged extension, if the extensions called do not answer.  This is where you will define it as per the illustrations below:

To do this, select the Follow me option; Setup -> Follow Me

You will be presented with the following screen:

Select the extensions that you want to define (the extension selection is on the right of the screen).  In my case, I picked my extension – ‘Ben Sharif <2001> add’

 

  In the screen that follows (see the illustration above), I entered the following information.

Every time my extension is called, Asterisk will try to connect to extension 2001 and if no answer, it will call my mobile.  If still no answer, it will drop to my mailbox.

 

3.6  Ring Groups

A ring group is a group of extensions that will ring when there is an external incoming call.  You can even put your Mobile Phone number in the ring group if you want to. The 0400123456# is my mobile phone (see illustration below).  For mobile phone to wok, you must have the appropriate route and trunk set up.

You may not want a ring group – it’s entirely up to you.  If you don’t require a ring group, you may ignore this section.

When there is an incoming call, the phones nominated in the selected group will ring.  You may select different ring group for each of the incoming trunk or you may nominate the same group for all the trunks, in which case you will only need to define only one ring group.

For simplicity, I have only defined 1 ring group for all incoming calls from all trunks – at this stage, let’s not get too fancy J

I created a ring group 10 for this purpose. I called it AllPhones as it rings all the phones in the group.  A single digit ring group is not recommended.

The ring group screen is illustrated below:

 

3.6.1  Now it’s a good time to set up your softphone.

To do this, go to the chapter – Setting up Soft Phone and come back to the next chapter after that is done.  

If everything has been done as explained above, you should be able to make and receive calls between your internal extensions.  If not, it is time to re-inspect what we have done above and make the necessary correction before attempting to go any further.

Let’s take a break and test the soft phone extensions by making calls to each extension.