Asterisk
(VoIP)
1. Overview
Telephone communication has not changed
radically since the telephone was invented in the late 1800s. New technologies
like digital circuits, DTMF (or, "touch tone"), and caller ID have
improved on this invention, of course, but its basic functionality still
remains the same. Over the years, service providers made a number of innovative
changes in order to improve the kinds and types of services offered to
subscribers, including toll-free numbers, call-return, and call-forwarding,
etc. By and large, users did not know how these services worked, but they did
know two things: the same old telephone is used, and the service provider
charges for every incremental service addition introduced.
In the 1990s, a number of individuals in
research environments, both in educational and in corporate institutions, began
to take a serious interest in carrying voice and video over IP networks,
especially over corporate Intranets and the Internet. This technology is
commonly referred to today as Voice over Internet Phone (VoIP). It is the
process of breaking up audio or video into small chunks, transmitting those
chunks over an IP network, and reassembling those chunks at the far end so that
two people can communicate using both audio and video.
The idea of VoIP is certainly not new, as
there are research papers and patents dating back several decades; and
demonstrations of the concept have been given at various times over the years.
VoIP took center stage with the "information super highway" (or, the
Internet) concept that was popularized by former US Vice President Al Gore in the
1990s. It was envisioned that the Internet would make it possible to
interconnect every home and every business with a packet-switched data network.
Before Al Gore's effort to grow the Internet, the Internet was generally
limited to use in academic environments only, but the possibility of mass
deployment of the Internet sparked renewed interest in VoIP.
2. Introduction
Asterisk© is a
Linux-based IPBX application developed by Mark Spencer of Digium™, the company
behind Asterisk. Asterisk@Home© evolved from the core of Asterisk. It is made
up of several major components. These were developed under the GNU General
Public License (GPL), supported relatively by the users themselves. It consists
of applications, a provisioning system, an installer, and an operating system
that, together, constitute a complete package ready-for-use as an
out-of-the-box PBX.
For the purposes
of this tutorial, the major components of Asterisk@Home© include:
1. Asterisk, the core PBX
2. Flash Operator Panel, a screen-based operator’s console
3. Asterisk Management Portal (AMP), a web-based provisioning tool for
Asterisk
4. A report system - that part of the AMP which provides CDR reporting
tools
5. A maintenance system, also a part of the AMP, which provides low-level
interfaces to some components, and real-time system information
6. CentOS, a version of Linux related to Enterprise Linux (but without the
branding and the support)
2.1. The Components
Two main
components need to be set up:
- An Asterisk-powered IP PBX
- The phones (or softphones)
- Network
2.1.1. The IP PBX
A dedicated PC is
needed in order to run an IP PBX. The PC described below is sufficient enough
to power an IPBX in a small office environment:
- 600 Mhz Pentium III PC or better
- 256 MB RAM – minimum
- 10 GB hard disk space - minimum
- 10/100 NIC
- CD-ROM Drive
- 10/100 4 or 8 ports Ethernet switch
2.1.2. Phones
You can use soft
phones as well as hard phones from Planet, and Cisco, etc.
To get started, it
is easiest to get a softphone, and run it on another computer. For details, see
the section on how to install a softphone.
2.1.3. Network
A static IP
address is needed to run Asterisk on a network (e.g. 192.168.0.12).
3. Installation and Configurations
Download the ISO from http://sourceforge.net/projects/asteriskathome/
- Burn the downloaded ISO image onto a blank CD.
- Ensure that your PC will boot from the CD. If necessary, change the BIOS settings to reflect this. Warning: This will erase all the data on the hard drives of the PC. If you have two drives, both may be erased as well - beware.
- Boot your Asterisk PC with the CD in the CD drive, and press the “Enter” key.
- Press the “Enter” key to start the installation. It will take approximately 30 minutes for the installation to be complete ready for the configuration stage.
- During this stage, you will see screens similar to the following. Linux and the required files are being installed. Wait for the process to end.
- After Linux is loaded, the CD will eject. Take the CD out, and wait for the system to reboot. You will then be presented with lines and lines of code. This process will take a while because it is building Asterisk.
Change Default Settings
Once Asterisk@Home
has been installed, some default system changes need to be made to Asterisk.
Log in to your new
Asterisk@Home box (user: root, password: password)
To
get help, At the command-line, type:
help-aah
A help list will
be displayed:
Change the Linux Password
Note: Change your root password immediately by typing:
passwd
Change some of the other default passwords as well; set your date, and
the IP address of your box to a static address.
Change the IP Address
Change the Asterisk IP address from DHCP to static. At the command
prompt, type:
netconfig
Select [Yes] to set up networking, and hit
the “Enter” key.
Use the “Tab” key to switch between various fields. Enter the IP address
that is to be allocated to the Asterisk box, the Netmask (sub-net mask),
Default Gateway and Primary nameserver, as per the example given above.
Select “OK”, and reboot the system.
Set Time Zone
Set the correct time zone. The following examples depend on whether you
are setting up AAH 1.x or AAH 2.x.
For AAH
1.x type the following at the command-line:
redhat-config-date
You will
get the screen on the left; choose an appropriate time zone. Check the system
clock. Use UTC if you want to.
Choose
“OK” to complete the process.
|
 |
For AAH 1.5 and 2.x, type the following at the command-line:
config
You will get the prompt as per the screen on the right.
Press D.
You will then be presented with the time zone screen.
Choose your time zone to complete the process.
|
 |
Log off Linux and reboot. Asterisk will now start with the new IP
address.
4. Connect to AMP from a Web Browser
Connect to http://ipaddress/ (e.g.
http:192.168.1.7) to configure Asterisk@Home. You will be presented with
an Asterisk Management Portal (AMP) as illustrated below:
4.1 Logging into an Asterisk Management Portal (AMP)
To log in to an Asterisk Management Portal (AMP) use the user: maint, and the password: password, unless you have
changed the password during the initial set up.
Once you have
logged into an AMP, you will be presented with the following screen:
Select the Setup tab and you will be presented
with the following screen.
This is where you
start configuring Asterisk@Home. Notice the selection options on the left hand
side. Selecting each option will display a configuration screen for that
particular function e.g. creating new extensions, creating new trunks, etc.
4.2. General Settings
Select General
Settings, and set them up as illustrated below:
Hover your mouse
over the corresponding field description. An amber underline will display the
purpose of the fields.
In the Asterisk Dial command option,
customize your preferences regarding Asterisk’s “behavior”, e.g. if you want
the caller to hear music instead of the standard ringing sound, replace the “r” with an “m”. For further options, hover your mouse over the label, and you
will be informed about other options. After setting up the General Settings,
click on the “Submit Changes” button, and the red bar on the top of the screen
for the changes to take place.
4.3. Extensions
The number of
extensions to be set up depends upon you. You can have soft phones installed in
four or five computers, or have a mixture of ATAs and SIP SoftPhones.
The following
extension numbers should be avoided:
200 - Park
notify
300-399 - Reserved
for speed-dial
70-79 - Reserved
for call-on-hold
80-89 - Reserved
for meet-me and conference
800-899 - Reserved
for meet-me and conference
8000-8999 - Reserved for meet-me and conference
80000-89999 - Reserved
for meet-me and conference
Select the type of
trunk e.g. SIP, IAX2, ZAP or Custom, from the Create Extension main menu
illustrated below:
The AAH v2.x’s
screen, where you actually create an extension, is given below:
Click on the “Add
Extension” button to add more extensions, e.g. 201, 202, 2000 and 2001.
Allocate the
password in the numeric form. If you enabled Voicemail, allocate the same
password as well. You may also nominate an e-mail address for Voicemail e-mail
notification.
5. Setting the soft phone
Now that Asterisk
is up, you need to setup up a soft phone. The following soft phones can be used
with Asterisk:
12.
SIPPS
softphone
15.
X-Lite
softphone
We will use the
BOL SIPPhone. It is extremely simple to set up for use with Asterisk, and it
also features a call-forwarding facility.
Obtain a copy of
the BOL 2000 SIPPhone from http://www.bol2000.com/download/sipphone/
After
downloading and set up, the following screen will be displayed when it runs:
5.1. Profile Tab
This is the only
screen that is required to be filled in.
Account: <enter the
extension number e.g. 201>
Password: <enter the
password e.g. 201>
Domain/Realm: <leave it blank>
Proxy: Your Asterisk network address e.g.
192.168.1.7
Port: 5060
Check the Auto Login and Keep Password.
Click OK.
5.2. Audio and Video Tab
Click on the
“Audio & Video” tab to ensure that the audio properties set is consistent
with the audio card installed in your PC/Notebook.
- Click on the Tuning Wizard to tune your audio input and output.
- Check Auto Send Video, if you are using video. Check it anyway.
- Check Auto Receive Video, if you are using video. Check it anyway.
Click OK.
5.3. Network Tab
Ensure that your
Internet Connection Type is set to LAN.
Ignore the
“Information of Network” field.
Click OK.
5.4. Call-forwarding
To forward an
unanswered call to an extension, click on the “Call Forward” tab, and enter the
telephone number you want to forward your incoming calls to. You have three
options of call-forwarding – Always On, Busy, or No Answer. This facility is
only available, however, if your PC is on, and the softphone is active.
Click OK.
You might want to
set-up a couple of PCs with the softphone, after which you can start testing
your new phone system by dialing each extension in turn.
If you use one of
the softphones and dial 7777,
Asterisk will simulate an incoming call.
Once done, test your
softphone connection to Asterisk.
To check if Asterisk is running
Click on the “System Status” tab. You will be presented with the
following screen:
5.5. Flash Operator Panel (FOP)
The Flash Operator Panel is a switchboard-type
application for the Asterisk PBX. It runs on a web browser with the Flash
plug-in. It is able to display information about your PBX activity in
real-time. The layout is configurable (button sizes and colors, icons, etc).
You can have more than 100 buttons active per screen.
It also supports contexts: you can have one server
running, and many different client displays (for hosted PBX, different
departments, etc).
It can integrate with CRM software, by popping up a
web page (and passing the CLID) when a specified button is ringing.
The following information is displayed on the FOP:
- Which extensions are busy, ringing, or available
- Who is talking and to whom (CLID, context, priority)
- SIP and IAX registration, and reach status
- Queue status (the number of waiting users)
- Message Waiting Indicator and count
- Parked channels
- Logged-in agents
Functions you can
perform on FOP:
- Hang-up a channel
- Using drag-and-drop to transfer a call
- Initiate calls by drag-and-drop
- Barge in on a call using drag-and-drop
- Set the caller ID when transferring or originating a call
- Automatically pop-up a web page with customer details
- Click-to-dial from a web page
No comments:
Post a Comment