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  1. #1
    Registered User
    Join Date
    Dec 2006

    SIP stack error 480


    I am trying to write a “Push To Talk Over Cellular” application, using the 3rd Edition of Symbian.

    When I send a plain SIP invite from user1 to user2, everything works fine, ie the Symbian SIP stack of the receiving client interprets the incoming invite properly. The invite in this case is routed through the SIP server.

    Now, I m trying to send a PoC invite to a group. In this case the SIP invite is routed through the PoC server.
    The problem that I am facing is that the Symbian SIP stack of the receiving client cannot interpret the incoming SIP invite which is now routed through the PoC server. And I get an error 480 : User Temporarily Not Available at the sending client. This happens even if the remote user is already registered. The incoming invite at the receiving client is :

    INVITE sip:kanchan@;transport=udp SIP/2.0
    Via: SIP/2.0/UDP;branch=z9hG4bK84.50bf.2bPIMK_STARTsip%3Akanchan%40ptalk.com%23%3Csip%3Asastry*test%40172.17.40.176%3A5080%3E%23333175906%23PIMK_ENDSTACK_TOKEN_START*1290248544*1097928232*2259943421T*98241237*STACK_TOKEN_END
    Via: SIP/2.0/UDP;branch=z9hG4bK45.4ae1.4
    To: "kanchan" <sip:kanchan@ptalk.com>
    From: <sip:sastry*test@group.ptalk.com>;tag=e67b1c-6784.2
    Call-ID: 4823-6bd38-291@
    CSeq: 6334 INVITE
    Max-Forwards: 69
    User-Agent: PoC-serv/1.0 Ericsson-PoC-Reference-Server/CXC1328363R1A011
    Record-Route: <sip:recordRouteSwapParameterPIMCONTACTSTARTsip%3Akanchan%40172.17.40.153%3A5060%3Btransport%3DudpPIMCONTACTENDPSTARTsip%3Akanchan%40ptalk.comPENDSTACK_TOKEN_START411372552STACK_TOKEN_END@ptalk.com:5070;maddr=;lr>
    Contact: <sip:sastry*test@>
    Content-Length: 275
    Content-Type: application/sdp
    Supported: timer
    Session-Expires: 1800
    Accept-Contact: *;+g.poc.talkburst="TRUE";require
    Allow-Events: refer, group-event
    Referred-By: <sip:sastry@ptalk.com>

    o=username 1035001843 1035001843 IN IP4
    s=SIP Session
    c=IN IP4
    t=0 0
    m=audio 35002 RTP/AVP 109
    a=rtpmap:109 AMR/8000
    a=fmtp:109 octet-align=1;mode-set=0,1,2,3,4,5,6,7

    Can anyone help ???????????

  2. #2
    Registered User
    Join Date
    Apr 2007

    Re: SIP stack error 480


    I am facing a problem in developing sip cleint for s603rdedition,
    My application ia able to send a INVITE request to another party phone.

    But when a INVITE request is comming on my client It have not receive that
    INVITE actually SIP stack take this message and send 480 responce to another party it have not send this INVITE to my application client.
    plz tell me why my application have not able to receving INVITE.
    I have already implemented call back function for that.

    If u have any plz send me

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