In this article, I will be demonstrating integration of Lync Server 2010 with Asterisk open source voice over IP solution. Asterisk can serve as gateway for Lync server in test environment for validating voice connectivity and feature.

Scenario:

ConfigurationTo help you setup direct SIP with Asterisk, step-by-step instructions for configuring Asterisk, Lync Server and the 3CX SIP client for Asterisk are discussed in this section.

A. Configuring Asterisk

Installation of Asterisk server is not discussed in this article. You can start after the default installation and basic “setup” is completed.

I am using Asterisk Graphical Interface to create a new Dial plan and two user extensions associated with this dial plan.

1. Login to Asterisk Graphical Interface “http://<AsteriskServerIP:8088/static/config/index.html”

2. Login with “Admin” user. In the left hand side menu expand “Dial Plans” section and click on “New DialPlan” button.

In “Create new Dial Plan” type the name of Dialplan in “DialPlan Name” Click Save.

3. Now Expand Users and click on “Create New User” Button.

Now add the details for new user

I have added two extensions for testing purpose

4. You can install any SIP phone client on desktop or from Android Market on Android phone. For this demonstration I have installed 3CXphone on my Android phone

Create extension profile on SIPPhone, Click on Settings button

Add details for the user we just created in Asterisk, to profile

Click on Save button to save the profile.

SIP Phone is now connected to the server you would be able to make calls to local Asterisk extensions

5. Let’s make a call to another extension and track the call status.

Run command below to open asterisk command line on Asterisk server

After running command, you will see the asterisk command prompt

To Show Current peer connections on Asterisk server

Dial extension 6001, which is having voice mail enabled, extension is not online so call will be routed to

voicemail. Enter the extension number in SIP phone and press

Call status will be visible in Asterisk command prompt

Recording sound starts because extension is not online

Configuring Asterisk for Lync integrationTo configure Asterisk, you must edit a series of configuration files. The following files are in a text format and are normally found in the /etc/asterisk directory:

sip.confextensions.conf

SIP.conf FileThe sip.conf file defines all SIP configurations for Asterisk. The first section in this text file is labeled [general]. In the [general] section, you define general SIP settings for the entire Asterisk server.

Next, we define a section in the sip.conf file to instruct Asterisk hzzzsow to connect to the Lync Server 2010, Mediation Server.

[Lynctrunk]   Tells Asterisk this is the start of a new section in the sip.conf file. Note the name is inside square brackets. You can name your SIP object anything you like.

type=friend   Set this option to friend. This tells Asterisk that this SIP object (that is, Mediation Server) is capable of sending and receiving calls.If this field is set to peer. This indicates to Asterisk that this SIP object can receive calls.

qualify=yes   Instructs Asterisk to verify that this SIP object is reachable. Asterisk performs a check every 60 seconds.

Extensions.conf FileThis file defines the dial plan configuration in Asterisk. The dial plan dictates how calls flow in Asterisk. Every incoming call that Asterisk receives is processed depending on the instructions defined in the dial plan.ContextsContexts are nothing more than a convenient container for grouping extensions.

After changing configuration reload Asterisk by using command

B. Configuring Lync Server 1. Using Lync 2010, Topology Builder define PSTN gateway

2. PSTN Gateway properties

3. Associate PSTN gateway to the Mediation pools and define listening ports:

4. Publish topology changes,

5. Configure Dial Plan, voice policy and route to define route for Asterisk Extensions

Open Dial plan OCSUMDial.ng.com to Create a Normalization rule.

Click New in “Associated Normalization Rules”

Add configuration and click OK

Click on OK,

Click on Voice Policy Tab and double click on default “Global” policy to open

Click On New on “Associated PSTN Usages”

Add “Name” of PSTN Usage record name and Click on New on Associated Routes

Add Route details and click OK

Click OK twice.

Route and PSTN Usage tab will populate with configuration we just added.

Commit the changes

Click Commit

C. Testing Voice Integration 1. Dial Asterisk Extension from Lync Client

Call will land to the Extension

When Call is answered

2. Dial Lync Extension from Asterisk SIP Client

Call is connected