Welcome to my Asterisk SJPhone Configuration Page. Asterisk is an open source Software based PBX that runs on Linux. Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world.

SJPhone is a SIP/VoIP Softphone for Microsoft Windows, This page is all about configuring Asterisk to work with SJPhone

If you have any questions or comments, please feel free to contact me by clicking the Contact Me link above.

For additional Asterisk information you can check out My Asterisk Resource Pages or optionally the Wiki
Step 1: Configure Asterisk
Configuring Asterisk to allow SJPhone to connect to it is done in two configuration files, the sip.conf file is what configures authentication and options for SJPhone to connect to asterisk.

The second file is your extensions.conf. To keep things simple, I am going to assume you are either running asterisk with the supplied sample configuration. If you have made changes to your Asterisk configuration files, you will need to adjust the following conf file examples accordingly.

This is from my /etc/asterisk/sip.conf. Generally you should only need to change the username and password portions to match what you've chosen. The username and password you select here will also be what you enter in SJPhone later.
[egg]                           
type=friend                     
host=dynamic                    
dtmfmode=rfc2833                
username=egg                  
secret=password
canreinvite=no                  
reinvite=no                     
callerid="Jim Radford" <6000> 
disallow=all                    
allow=gsm

Now here is a context from my /etc/asterisk/extensions.conf that is used with this particular sjphone installation. This information should be placed in the demo context somewhere, or if placed in another context it should be included in your default context. If this doesn't make any sense, just add the following lines somewhere in the [demo] context.
exten => 6000,1,Dial(SIP/egg,10)
exten => 6000,2,Hangup
exten => egg,1,goto(6000,1) ; To be able to dial with text, "egg"
What this does is allows anyone connected to asterisk to dial either "egg" or "6000" to ring my sjphone, if I don't answer in 10 seconds it proceeds and hangs up the connection.
Step 2: Configure the SJPhone Client

You will need to download a copy of the SJPhone Client from SJLabs Here. This Document was written using SJPHone 1.30 Dated 25-Oct-2004

1. Start SJPhone and then click on the Options Icon, Optionally you can right click in the screen display area and click options to access the following setup screens.


2. Under the User Information Tab, You can put your name, location, and Email address.


3. Under the Call Options Tab, You can configure SJPhone to automatically accept calls when your extension is dialed, I do this sometimes when testing from a different computer.


4. In the Profiles Menu, click "New" to create a new profile for your asterisk server



Enter your profile name, and select "Calls through SIP Proxy", Then click OK
Back at the Profiles menu now (screenshot above), go ahead and click your newly created profile name, then click "Use".

Now, lets go ahead and click Initialize, the Service: dialog will pop up, Put your sip.conf username and password here. This is the same information you used to configure the sip.conf file in Step 1.


Click on the Edit Button so we can configure the asterisk server information. The first screen is the Initialization screen, these are the default settings I use. Click on the SIP Proxy Tab.


Under Proxy Domain, put the IP Address of your Asterisk Server and make sure Register on proxy is checked.


Configure your general settings tab options to match the ones shown below


I had trouble when I first installed SJPhone and it was because I had checked the Use Discovered Addresses in SIP. If your settings match mine you should be Ok.


5. These are my ILS server settings, I don't run one, and since I use Asterisk for SIP termination I don't need one. Your settings should match mine


6. Here are the audio options I use, if these do not work for you, the defaults probably will, so be careful about changing them. I changed no options under Advanced Settings or Compression for my setup.

7. The following screenshots are here so you can see my configuration, none of the settings below should make a difference with your ability to connect to asterisk.

You can click OK and skip to Step3 now if you wish.

Step 3: Communication with Asterisk

Now that you've gotten your configuration working, the screen on your SJPhone should say something simuliar to mine shown in the screen shot at the beginning of Step 2.

Since I assume you're running with the demo setup files, you should be able to dial 1000 to reach the demo main menu, or from the console (or even your own SJPhone) dial 6000 to ring yourself.

If you have any problems, go back through the screen shots and be sure your settings match mine other than username and password. Use the username and password you configured in your sip.conf file in Step 1. Be sure to check the Asterisk Console for debugging information.