Linksys spa-3102

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Revision as of 08:39, 10 June 2008 by Langstonius (Talk | contribs) (Configuring FreePBX)

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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