Revision as of 08:41, 10 June 2008

Linksys SPA-3102

Specifications

The Linksys SPA3102 features the ability to connect standard telephones and fax machines to IP-based data networks with the additional benefit of an integrated connection for legacy telephone network "hop-on, hop-off" applications.

• IP Telephony features
  • VoIP Protocols SIP v2
  • Voice Codecs G.711, G.723.1, G.729a, G.726
  • Telephony Interfaces 1 phone (FXS), 1 line (FXO)
  • IP Telephony Features Echo cancellation (G.168)

Setting up with linuxMCE

Configuring the Device

• Step 1 - Initial setup.
  • Connect the SPA-3102 to your internal network (the one connected to the lmce subnet)
  • Proceed with initial configuration via the administration page. Once you set passwords and locale settings, you can move to the next step
  • Note, you can leave the dhcp and network settings as is. We will not be using the routing functions of this device. You will need to make a note on the address assigned to the device.

• Step 2 - SIP connection settings.
  • Click the PSTN Line settings.
  • Choose 'Admin' and 'Advanced' option in the top left right corner to see all available options.
    • Do not change anything other than what you need. There are a lot of options!
    • Line Option Set to enable.
    • Sip Settings
      • Make sure the port is set to 5061
    • Proxy and registration
      • Proxy: the ip address of your core.
      • Register: Yes.
      • Make calls without registration: - Not tested but set to yes.
      • Answer calls without registration: Not tested but set to yes.
    • Subscriber Information
      • For Display Name I put 'Landline'.
      • For user id I put the extension I created for this line.
      • For password I put the password for the extension
      • For 'Use Auth Id' I chose no.
    • Dial Plans
      • Under Dial Plan 2 add this '(<S0:600>)
    • PSTN-To-VoIP Gateway Setup
      • PSTN-To-VoIP Gateway Enable: Yes
      • PSTN Ring Thru Line 1: No
      • PSTN CID For VoIP CID: Yes
      • PSTN Caller Default DP: 2
        • This is the dial plan we changed earlier.
        • You may need to change some of the other items such as FXO timing, but thats beyond this article for now.

Configuring FreePBX

• Step 1
  • Log into the webadmin.
  • Using the top navigations, goto Advanced -> Configuration -> Phones Setup.
    • This will take you to the freePBX panel. Take care to only change designated settings as you can mess things up.
  • On the left hand side, choose Trunks from the menu.
  • This will show you the existing trunks (phone lines). Choose Add Trunk.
  • Then choose Add Sip Trunk.
  • Next are the options presented:
    • Outbound Caller ID : Caller ID for calls placed out on this trunk.
    • Maximum channels: Set this to '1'
    • Dial Rules : Still working on this part. Idealy, it would be used for local non toll calls as well as 911 or any other local services.
  • Outgoing Settings

 allow=ulaw
 canreinvite=no
 context=from-pstn
 disallow=all
 dtmfmode=rfc2833
 host=192.168.80.233
 incominglimit=1
 nat=never
 port=5061
 qualify=yes
 secret=xxx <----extension password
 type=friend
 username=xxx <----username on the spa-3102 that you set should be pstn

• • Incoming Settings
    • This is currently empty.