Making calls by attaching Google Voice Trunk to Asterisk Server TSMN 6350 IP Telephony: Final Project Presentation December 5, 2011 Team Members-Ashok Galla, Dinesh Reddy Attunuri, Saloni Shah

Introduction to Google Voice

Network Diagram IP PBX Internet PSTN Router IP Phone

Attaching Google Voice trunks to Asterisk Server : Demo

Objectives  Build a server with required configuration to support Asterisk.  Install Asterisk on the configured server.  Install Free PBX.  Add custom Google Voice trunks.  Making extensions.  Configuring the.conf files  Test the interoperability with soft sip clients and an IP phone.

Attaching Google Voice trunks to Asterisk Server : Results 1. Building server with required configuration: Dell Optiplex GX280 Intel P4 2.4Ghz, 6Mb cache, 2 GB RAM 2. Install Trixbox (based on Asterisk)on the configured server: To support google voice on asterisk, we need a minimum version of 1.8. Asterisk communicates with Google Voice using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, we ensured that both are compiled and part of our installation.

3. Install Free PBX : Installing FreePBX was necessary because it is the primary management interface for managing users, extensions, ring groups, queues, trunks, and more. This is where we spent the majority of our time configuring the system. FreePBX offers better reliability and performance that enabled us to improve productivity and reduce operational costs (It’s actually free).

4. Add custom Google Voice Trunks: We need a dedicated Google Voice account to support calls using FreePBX. So we created one. The details of custom google trunk are as below: Dial Rules: 1+NXXNXXXXXX but if your area allows seven digit dialing, you may also want to add one like 1areacode+NXXXXXX where areacode is replaced with your local three digit area code. Custom Dial String: We set it to voipuser6

The res_jabber Resource is configured with the jabber.conf configuration file, typically located in /etc/asterisk.

We configured our dialplan to receive an incoming call from Google and route it to the SIP phone we created. To do this, our dialplan, extensions.conf, typically located in /etc/asterisk, would look like:

6. Test the interoperability with different soft sip clients in different environments: We used several soft sip clients like Zoiper Communicator on a Mac and X-Lite on Windows.

Scope  The only drawback to Google Voice's free U.S. and Canada calling with the Free PBX has been the fact that we could only make one outbound call at a time.  Future scope of the project is to add more than one custom trunk to enable more than one simultaneous inbound and outbound calls as a solution to setup a VoIP network for small business organizations.  Skype enables video. Next step from this project would be to make video calls using a video enabled IP Phone by attaching Skype Video trunks to Asterisk.

Wireshark: Results

The END  Questions?  Comments?  Applause?  Tomatoes?