Setting up test users(sip) in Asterisk

Setting up test users(sip) in Asterisk

Two files to modify

-sip.conf

-extensions.conf

Software needed:

Twinkle for Ubuntu 11.04

http://mfnboer.home.xs4all.nl/twinkle/download/twinkle-1.4.2.tar.gz

Ekiga for Windows (7)

http://ekiga.net/yannick/win32/ekiga-setup-3.2.7.exe

Of course the softphone can be anything you choose but this is what im using

This is a demonstration of a call between a Ubuntu sip user and windows user who both can connect to the Asterisk server as set up in previous steps. The GUI can be used for this,but as recommended getting to know the files personally is a lot better in the long run.

Ok so here goes:

Adding sip users

open ter

sudo su root

navigate your way to /etc/asterisk

vi sip.conf

Only add the text in bold

[test]

--user register

type=friend

--user can make or recieve calls peer =inbound only, user = outbound only

username=test

--username

secret=password

--password

host =dynamic

sets dynamic ip for the host

context=tutorial

sets the group test will belong to.

Our second user

[test1]

type=friend

username=test1

secret=password1

host =dynamic

context=tutorial

Adding the users to the extensions

navigate your way to /etc/asterisk

vi extensions.conf

[tutorial]

exten => 1234,1,Dial(SIP/test)

exten => 4321,1,Dial(SIP/test1)

exten => number,prioirty,Dial(protocol/user)

Now asterisk -r

reload

sip show peers

The users should now be there!

Adding the softphones

Twinkle for Ubuntu

Install twinkle

wizard

enter name for profile

user name test

domain 192.168.0.23 –this is what the domain was for me,so whatever your ip address is that the asterisk server resides on

authentication name test

password password

Ekiga for windows 7

Once installed

Accounts->Add a sip account

name test1

Registrar 192.168.0.23

user test1

authentication user test1

password password

enable account (ticked)

To test the call functionality

From either machine

(ubuntu in this case and twinkle)

make sure user is registered(logged in)

Call 4321

Hopefully it will come up as an incoming call on the ekiga machine :-)

From windows machine using Ekiga

sip:1234@192.168.0.23

hit call icon

To see the communication between both login into the asterisk server

asterisk -r

This was done on my home network so modifications for accessing the asterisk server from outside the home network will be done soon

Update: 
The ip for the registrar is the outside ip address of the server which you are using. provided that port forwarding is enabled and working properly then connecting to the asterisk server should not be a problem. From a clean install on a VM outside my network I was able to connect and register the SIP user without having to change any network configuration files for asterisk.