[Linksys 3102 + GrandSteam IP phone] Direct outgoing IP call?

Discussion in 'VOIP' started by Vincent Delporte, Sep 4, 2006.

  1. Hello

    I've successfully set up the Linksys to ring up a GrandStream
    IP phone directly (ie. with no SIP PBX acting as registrar) when it's
    installed on the LAN, but the Linksys fails dialing that IP phone
    through the Net.

    Both the Linksys and the IP phone are located in private networks with
    NAT firewalls standing on both sides, so I imagine I must modify some
    settings in the Linksys, but I have no idea what to try besides adding
    a STUN server in the SIP section (tun.fwdnet.net:3478, and STUN Enable
    = Yes), and also configuring up the IP phone to use "NAT Traversal"
    and the same STUN server as the Linksys.

    FWIW, here's the output when I dial into the Linksys through the PSTN
    port (the Linksys uses IP 192.168.0.253, 087077XXXX is the caller ID
    number, sip.acme.com is the fixed public IP through which the IP phone
    can be reached):

    --------------- BEGIN LOG ------------------
    <151>FXO:Start CNDD

    <151>FXO:CNDD name=, number=087077XXXX

    <151>FXO:Stop CNDD


    <159>FXO:CNDD Name= Phone=087077XXXX


    <151>AUD:Stop PSTN Tone

    <151>AUD:Stop PSTN Tone

    <151>Calling::5060

    <151>[1:0]AUD ALLOC CALL (port=16394)

    <151>[1:0]RTP Rx Up


    <159>RSE_DEBUG: reference domain:sip.acme.com


    <151>RSE:GetServerAddrErr(sip.acme.com,0)=-101

    <151>TP:?Tx->0


    <134>[1]->0.0.0.0:5060

    <134>[1]->0.0.0.0:5060

    INVITE sip::5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.0.253:5060;branch=z9hG4bK-5924bcd9
    From: fxo <sip:087077XXXX@192.168.0.253>;tag=c0823086953a9751o1
    To: <sip::5060>
    Remote-Party-ID: fxo
    <sip:087077XXXX@192.168.0.253>;screen=yes;party=calling
    Call-ID: 4e39ec6a-a7c16cbd@192.168.0.253
    CSeq: 101 INVITE
    Max-Forwards: 70
    Contact: fxo <sip:087077XXXX@192.168.0.253:5060>
    Expires: 240
    User-Agent: Linksys/SPA3102-3.2.10(GW)
    Content-Length: 277
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp

    v=0
    o=- 19360 19360 IN IP4 192.168.0.253
    s=-
    c=IN IP4 192.168.0.253
    t=0 0
    m=audio 16394 RTP/AVP 0 8 100 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:100 NSE/8000
    a=fmtp:100 192-193
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv

    <134>

    <134>


    <159>[0]FM Alert Stop RxTx (c=0023d298;a=0)


    <151>[1:0]AUD Rel Call

    <151>CC:Failed w/ Calling

    <151>AUD:Stop PSTN Tone


    <159>RSE_DEBUG: unref domain, sip.acme.com

    <159>RSE_DEBUG: last unref for domain sip.acme.com


    <151>Sess Terminated

    <151>AUD:Stop PSTN Tone

    --------------- END LOG ------------------

    If someone successfully set up the Linksys to call out an IP phone
    directly, ie. without first going through a VoIP server, can you tell
    me what to change?

    Thank you.
     
    Vincent Delporte, Sep 4, 2006
    #1
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