Accessing Exchange 2007 Unified Messaging: Part 5 – Configure the SIP client

Now that we have configured the Exchange UM Services and the basic sipX services, we can test to see if our configuration is working. In fact, if all you need is soft phone (SIP) access to the Exchange Server, then you don't need to configure the Asterisk server. Forward the appropriate ports on your router to the sipX server instead. The next part of this guide, Configuring Asterisk, will allow us to dial into the Exchange Server from a PSTN or a SIP phone.

You can use any SIP compatible client, but for our testing, we will use X-Lite from CounterPath. X-Lite is the freeware version of CounterPath's EyeBeam soft phone. Both versions can be downloaded from

When you first start X-Lite, you are given the opportunity to setup a SIP account. Press the Add button and enter the information below.

Display Name: Your Name
User name: 300
Password: SIP Password you entered in sipX for the test extension
Authorization user name: Leave Blank
Domain: sipX.lithnet.local

Click OK on the properties page, and Close on the accounts list page. The information display in the X-Lite main window will show the process of registering with the server, and if all goes well, the display should say Ready as per the image below.

Now in order to test that what we have setup so far is working, dial extension 300. This tests that we have correctly registered with sipX for incoming calls. You should see an incoming call on line 2.

In order to test the connection to the Exchange Server, dial 222, and the UM server should pick up and say "Welcome, you are connected to Microsoft Exchange". As long as you set your extension to 300 when you enabled your mailbox from UM, Exchange will detect that you are calling from 300, greet you by name, and simply prompt for your PIN. Now you can test the capabilities of Outlook Voice Access and navigate through the menus. This great guide is available for download from Microsoft and shows you all the commands available for Outlook Voice Access -

The sipX server passes the User ID of the connected user (in our case, '300') to the Exchange server by means of Caller ID. If Exchange can match the Caller ID of the incoming call to an extension specified on a UM enabled mailbox, then it will assume that user is calling, and just ask for the PIN. If it can't match the Caller ID, then it will ask you for your extension number, followed by PIN.

If you get a 404 Not Found error when trying to dial 222, go back and check that the gateway and dial plans are correctly configured in sipX. Just to be sure, activate your dial plans again. This error tells us that either the Exchange Server rejected the call because it could not find extension 222 on its system, or that sipX rejected the call because a dial plan could not be found to route the call through.

If you get a 408 Timeout error, try dialing again. This is actually a better error to get than the one above. Provided you get the message when attempting to make a call, it generally means the sipX server was able to find a dial rule through which to route the call, but the gateway did not respond in time. See the troubleshooting section for information on capturing packets using Wireshark, and have a look at where the packets are being directed to.

To confirm that we have setup the auto attendant correctly, dial 299, and the UM server should pick up and say "Welcome to the Microsoft Exchange Auto Attendant. To reach a specific person, just tell me their name". If you speak the name of someone in the directory, it will try to call them. As we have not completed our setup, it will divert to voicemail. If you leave a voicemail, it will appear in your Outlook inbox. If OVA and the AutoAttendant are working, our sipX and Exchange configuration is complete, and we can move onto configuring the Trixbox server.

Next: Part 6 - Configuring Asterisk/Trixbox
Previous: Part 4 - Configuring the sipX Server