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Asterisk Basic Configuration


What is Asterisk?


Officially, Asterisk is an Open Source hybrid TDM and packet voice PBX and IVR platform with ACD functionality. Unofficially, Asterisk is quite possibly the most powerful, flexible, and extensible piece of integrated telecommunications software available. Its name comes from the asterisk symbol, *, which in UNIX (including Linux) and DOS environments represents a wildcard, matching any filename. Similarly, Asterisk the PBX is designed to interface any piece of telephony hardware or software with any telephony application, seamlessly and consistently.

Traditionally, telephony products are designed to meet a specific technical need in a network. However, many applications of using telephony share a great deal of technology. Asterisk takes advantage of this synergy to create a single environment that can be molded to fit any particular application, or collection of applications, as the user sees fit. Asterisk can, among other things, be used in any of these applications:
::Heterogeneous Voice over IP gateway (MGCP, SIP, IAX, H.323)
::Private Branch eXchange (PBX)
::Custom Interactive Voice Response (IVR) server
::Softswitch
::Conferencing server
::Number translation
::Calling card application
::Predictive dialer
::Call queuing with remote agents
::Remote offices for existing PBX

Perhaps more importantly, it can fill all of those roles simultaneously and seamlessly between interfaces.

Released versions of Asterisk can be freely downloaded from Asterisk 1.4.0 or Asterisk 1.2.14 via FTP. For more information, log on http://asterisk.org/

Asterisk Basic Configuration with WorldDialPoint

Asterisk Free PBX Screen Shots

Using standard asterisk v1.2.13

- This example is based on an Asterisk connecting to a Business Plan. Residential Plan users please login in your module and click on Hardware Configurations.

Here is what is required. - useragent=PAP2T

SSH into your box, and then edit (use nano) the file /etc/asterisk/sip.conf

Make the top part of the file read like this:

[general]

port=5060 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine)
nat=yes
disallow=all
allow=g729
allow=g723
allow=alaw
insecure=very   (Asterisk Version 1.4 change to - insecure=port,invite)
useragent=PAP2T



Once you have made that change, save the file and exit.

Then, in free pbx, all you need is:

Outgoing settings:

Trunk Name: WDP
Peer Details:
host=
203.176.185.10
secret=password
type=peer
username=login number


And incoming settings:

User Context: 619901*****
User Details:
context=from-trunk
secret=password
type=user


And, let's not be without registration:

login number:password@203.176.185.10/login number

Restart asterisk, and there you go. Ready to make Calls
Enjoy.

Call the number 09 9015 9002 to check your voice
 

Now that you are setup, you are ready to make and receive calls