Asterisk is an open source architecture which is sponsored by Digium for developing communications applications. Asterisk turns a standard PC into a communications server.Asterisk powers conference servers, VoIP gateways, IP PBX systems etc. It is used by both small & large businesses, carriers, call centers and by companies worldwide.
Download asterisk gui from below link:
File to be downloaded: asterisk-220.127.116.11.tar.gz
Install downloaded Asterisk GUI on your linux system. Below is a guide to install asterisk on linux systems.
After installing Asterisk GUI, you need to configure your asterisk.
We recommend you to take backup of your configuration files before editing them. To take backup just copy all file from /etc/asterisk under different location or name in the same directory.
cp –r /etc/asterisk /etc/asterisk_backup
We need to configure two files which will make asterisk work.
Edit manager.conf and add below configuration. We need to set enabled and webenabled to yes and need to add a new user to manager.conf.
enabled = yes webenabled = yes [admin] secret = password read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config
After making above modification save your manager.conf and edit http.conf file and add below configuration:
enabled = yes enablestatic=yes bindaddr=0.0.0.0
After making above modification save your http.conf. and then run below command in your asterisk-gui directory.
This command will check if asterisk GUI is configured correctly.
Restart your system. Then type following in address bar of your browser.
A login screen will appear, you can login using user credentials which we configured in our manager.conf ie. Username – admin & Password – password.
Note: In our case IP Address of our linux system is 192.168.1.133 which we will be using in our tutorial. You need to change it with your system IP Address.
You can use asterisk GUI to edit asterisk file or edit them manually.
Open /etc/asterisk/sip.conf in vi editor using linux terminal. Type following commands in terminal:
cd etc/asterisk/ vi sip.conf
Then bind address & port and add extension in sip.conf
[general] port = 5060 bindaddr = 0.0.0.0  type = friend host = dynamic secret = 1000  type = friend host = dynamic secret = 1001  type = friend host = dynamic secret = 1002
Save sip.conf by using following command:
The above rule in sip.conf will add extensions 1000,1001 & 1002 in asterisk.
After configuring your sip.conf, you need to create a dialplan in extensions.conf for enable calling between extensions created above. Open extensions.conf in vi editor and add below dialplan under default context.
Open terminal and type following commands
cd etc/asterisk/ vi extensions.conf [default] exten=1000,1,Dail(SIP/1000, 20) exten=1001,1,Dail(SIP/1001, 20) exten=1002,1,Dail(SIP/1002, 20)
Save extensions.conf by using following command:
Restart your asterisk server.
After configuring sip.conf and extensions.conf, configure the your hard phones & soft phones.
You can download XLite if you want to use softphone. Download URL: http://www.counterpath.com/x-lite.html
To configure your X-lite open System Settings > SIP Proxy > [Default] :
Set Enabled : Yes
Username : 1001 (which we have defined in sip.conf of asterisk.)
Authorization User: 1001
Password: 1001 (secret defined in sip.conf under 1001 context.)
Domain: 192.168.1.138 (Asterisk server IP)
SIP Proxy: 192.168.1.138 (For now asterisk server IP)
Outbound Proxy: 192.168.1.138(Asterisk server IP)
Now to configure your hard phone, in our case we are using GRANDSTREAM-GXP1405
Go to config section of your phone
SIP Proxy = 192.168.1.138(For now Asterisk server IP)
Outbound Proxy = 192.168.1.138(Asterisk server IP)
UserId = 1000
AuthenticationId = 1000
Password = 1000 (secret defined in sip.conf under 1001 context.)
The above settings will enable phones with configured extension 1000 and 1001 to call each other and to receive call if someone dial their respective extension.
Tags: adding user for asterisk gui, calls between extension asterisk, configure extensions.conf, configure hard phone, configure manager.conf, configure soft phones, configuring sip.conf, digium asterisk, x-lite