[Linksys 3102] Care to share a working config?

Discussion in 'VOIP' started by Vincent Delporte, Aug 24, 2006.

  1. Hello

    (FYI, I'm not using the WAN interface and the Linksys is connected to
    the LAN using the familiar plan, the PBX is sitting on
    the same LAN, and no firewall is involved. This is straight PSTN ->
    Linksys -> LAN -> PBX.)

    No matter what I try, I still have the following road-blocks:

    1. If I don't forward calls to the PBX through User 1 > Cfwd All Dest,
    Linksys doesn't notify the PBX of incoming calls. Syslogd does show
    that the Linksys detects the call. Using a dial plan under PSTN Line
    doesn't work

    2. When calls do get set, the internal extension can hear PSTN caller,
    but PSTN caller can't hear internal extension : routing problem?

    Since the 3102 doesn't come with any documentation and I'm told it's
    sufficiently different from the 3000 that I shouldn't rely on
    documentation for the latter... if some 3102 experts out there could
    take a quick look at my configuration and what I tried and see if
    there's something obvious missing/wrong, I'd be very grateful.


    Thank you.
    Vincent Delporte, Aug 24, 2006
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  2. (...which by the way is not done with the configuration you posted at
    I think that the softphone mistakenly sends its RTP packets to a wrong
    address or port. This could be due to some piece of equipment (e.g., the
    PBX) using STUN to get the router's external IP address, and declaring that
    one instead of the internal address (in the 192.128.0/24 network). This is
    common practice to handle calls with peers ouside the NAT, but in your case
    it would cause the loss of the RTP packets sent by the softphone...
    Unfortunately, it's hard to tell from a distance. Why don't you try running
    Ethereal (a.k.a. WireShark) to see what's going on? You may define a capture
    filter like "udp and not port 53", just to avoid most unrelated packets.
    Note: if you want Ethereal to sniff packets sent by the SPA-3102 to some
    wrong place, you must use a non-switching ethernet hub (good luck to find
    one in these days...). But in your case it would seem that it's the packets
    sent by the softphone which are going astray, so they should be well visible
    to Ethereal if it's run on the same PC.
    The voice part looks identical to me... If I were you, I'd grab
    http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf and study it
    carefully... The section 4.13 has various "Configuration Examples and Call
    Scenarios" that were quite useful to me.
    At first glance it looks fine, although as I said it looks consistent with
    the "second method" rather than with the one based on call forwarding. I
    can't understand why the call is not passed to the PBX: you even register
    the FXO on it...

    By the way, Voxilla.com at http://voxilla.com/spa3kconfig.php has a useful
    wizard to help in the configuration of SPA-3000 / SPA-3102 , which can at
    least preset a few defaults. It assumes the use of a service provider, but
    you can later adjust the server's name changing it into the IP address of
    your PBX.
    IIRC, you need to register before accessing it.


    Enzo Michelangeli, Aug 24, 2006
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  3. Finally found what it was:


    Linksys "support" is beyond words, and I couldn't bear one more day of
    staring at that interface ;-)

    Thank you very much for your help, though. It did turn out to be a
    routing issue.
    Vincent Delporte, Aug 24, 2006
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