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Asterisk is a open source VOIP PABX which is considerably powerful and nowadays many products are based on this. Following is a confguration which I tested in my development environment and it motivated me to dig in to voip world few years ago. This is the start and you will be amazed by it's features. I will stop from here and will leave it for you to experience...

Installation

1. Install a fresh copy of Linux (Red Hat Linux with Kernel 2.4 or higher is preferred). I installed it on a Red hat Linux 9.0 with Kernel 2.4

2. Download the latest source from www.asterisk.org

3. Copy the *.gz file to the /usr/src directory

4. Extract the *.gx file by using the following command

gzip asterisk-1.0.7.tar.gz

5. Now extract the Tape Archive (TAR) by using the following command

tar –xvf asterisk-1.0.7.tar

6. Now move to the asterisk-1.0.7 folder

7. Run the following commands to install Asterisk

make clean
make
make install


8. The above process will take sometime and will take nearly 10 minutes

9. In the meantime download a Soft phone such as X-Lite from

http://www.xten.com/

10. Install this on a Client Computer with windows

11. After the compilation and the installation is completed move to the following directory

/etc/asterisk

12. Open the following file by using your favorite text editor

sip.conf

13. Specify the following values (Trust me this is very easy to configure .conf file)

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ; this is important for incoming calls

You can have any word for context but make sure you use the same in other places (This is where you can go wrong)

14. Specify an authentication string

register => 21103:mypassword@mysipserver/1

21103 – Is the username which must be specified in the IP phone
Mypassword – Is the password which must be specified in the phones
Mysipserver – Is the hostname of the SIP server
/1 - is a bogus extension number

15. If you are using a SIP proxy which is used to connect to any other server edit the following section of the sip.conf

[mysipproxy.com]
type=peer
host=IP ADDRESS OF THE SIP SERVER
fromuser=<AUTHENTICATION ID>
secret=mypassword
fromdomain=<YOUR DOMAIN NAME>


16. Define you extensions at the bottom of the sip.conf file as follows

[phone1]
type=friend
host=dynamic
defaultip=192.168.1.4
;username=blah
;secret=blah
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip Note – This must be equal to the context you define on the top
callerid="Me" <2124>

[phone2]
type=friend
;secret=blah
host=dynamic
defaultip=192.168.1.3
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
mailbox=1000 ; Mailbox for message waiting indicator
context=sip
callerid="Mini Me" <2123>

17. Now save the file and open the extensions.conf file which is also in /etc/asterisk and add the following to the bottom of the file

[sip]

exten => 1,1,Dial(SIP/phone1,20,tr)
exten => 2,1,Dial(SIP/phone2,20,tr)
exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr)

Note – [sip] This should be the same context name used in the sip.conf file for incoming calls that’s why I said context is very important!!!

18. Upon completing the above start asterisk as follows from /usr/sbin

asterisk

or run it in debug mode as

asterisk –vvvvvvvvffffffffffff


19. If you need to enter asterisk SIP CLI enter the following

asterisk –r

20. This will take you to command prompt and by typing ? will provide you the available commands

21. That concludes my basic setup of configuring asterisk. This configuration worked with me. I will include the configuration steps of Voice mail, Call waiting and fine tuning later….. Now you can enjoy IP telephony by using the extension 1 and 2 !!! enjoy

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