Difference between revisions of "Linksys SPA3000"

Revision as of 16:26, 16 January 2009

General Information

The Sipura (now bought out by Linksys) spa 3000 is a pstn (analog phone line) to Asterisk gateway device that provides a SIP interface for 1 FXO port (analog phone line) and 1 FXS port (analog phone). This allows you to use the Telecom section of LMCE, and use Asterisk to send/recieve/route phone calls.

Most office environments have a phone in each room with their own extension number. Most homes, however, are set up with 1 common line (extension) that all phones are connected to - when a call comes in, all phones ring together. LMCE will allow you to have either setup (if you have enough FXS ports to support the number of extensions you want). Since the spa 3000 has only 1 FXS port, I am going to show you how to set it up so that all phones on the main house phone line will be treated as one extension and ring together. The picture below illustrates the installation method that I'm going to outline here.

Another interesting feature of the spa3000, is that if the LMCE network goes down, or if there is a loss of power, the spa3000 will bridge the FXO and FXS ports, bypassing LMCE and Asterisk alltogether, so that you will still have use of your phones in such emergencies.

Also, please realize that these instructions are for manually setting up the spa3000. In the near future, the process will be automated, and if this feature is available at the time you are reading this, then you should go the automatic route.

Setup

We are going to accomplish this setup in 3 steps: 1- Configuring the Phone Line in the web admin (FreePBX) 2- Adding the device for the internal phone line (FXS port) 3) Configuring the spa3000 (as we will need information from the previous 2 steps to complete)

Step 1 - Configuring the Phone Line

First, log into the web admin at 192.168.80.1, then navigate to Advanced->Configuration->Phones Setup. This will bring you to the FreePBX Admin page On the left hand side, click on "Trunks", then click "Add SIP Trunk" on the page the follows. You will now have to fill out the folowing information:

Outbound Caller ID: House Line

Never Override Caller ID: leave unchecked

Maximum Channels: 1 (this must be set to 1 as a pstn line can only handle 1 call at a time)

Disable Trunk: leave unchecked

Monitor Trunk Failures: leave the Enable checkbox unchecked

Dial Rules: Leave blank

Dial rules wizards: leave this alone

Outbound Dial Prefix: leave blank

Trunk Name: House Line (note: remember this! we have to match this in the spa3000 setup!)

Peer Details:

allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=ohio98yo
type=friend
username=1-pstn