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Thread: SIP/2.0 403 Non-self Request-URI

  1. #1

    Default SIP/2.0 403 Non-self Request-URI

    After an extended week-end, I started working this morning from home with my office line redirected to my sip line at home. Interestingly, all calls I got today were to my cell and many cut out after 19 seconds. So when I asked somebody to call my office number and my cell still rang I started getting suspicious and tried it myself, and wouldn't you know, all calls to my home were diverted to my call. This is supposed to happen if my sip line is not registered, but sure enough the Quadro reported everything was okay.

    In order to determine where the problem lies, I looked at the diagnostic logs and found something I have never seen before: calls still come in to my Quadro, but the Quadro turns them down with the error message in the title.

    Below I have one such incoming call in full detail. Now I figure my IPTSP screwed up yet again but I don't know how exactly and I'm wondering if anybody could tell me what they need to fix in order to get my line working again (to make things even better the IPTSP recently changed their web interface so I cannot take out the rollover to my cell anymore.. so now I have to pay a VoIP to Mobile call for every incoming call - so far I've already ratched up more than $10 bucks today alone).

    Code:
    18:38:31 Receive SIP message # (13/05/2008 16:38:31:457 GMT) # UDP # 1032 bytes # from: 91.121.101.126:5060 # to: 192.168.1.10:5060
    
    ***************************** SIP message buffer start *****************************
    INVITE sip:41225500341@213.221.210.139:5060 SIP/2.0
    Record-Route: <sip:91.121.101.126;ftag=7e82800a29a780c91375fb1c6b984d10;lr>
    Via: SIP/2.0/UDP 91.121.101.126;branch=z9hG4bKccc1.022038c2dd4cb31a9b016fbb0c6eb9d9.0
    Via: SIP/2.0/UDP 91.121.101.126:5061;branch=z9hG4bK7584bfd86b96500312c96c574f060abe;rport=5061
    Max-Forwards: 16
    From: <sip:+41764400351@91.121.101.126>;tag=7e82800a29a780c91375fb1c6b984d10
    To: <sip:41225500341@91.121.101.126>
    Call-ID: DA24E1F1-204111DD-BF2B857F-4FFC935D@146.188.127.1
    CSeq: 200 INVITE
    Contact: Anonymous <sip:91.121.101.126:5061>
    Expires: 300
    User-Agent: Sippy
    cisco-GUID: 3659770273-541135325-2164981780-473918838
    h323-conf-id: 3659770273-541135325-2164981780-473918838
    Content-Length: 262
    Content-Type: application/sdp
    
    v=0
    o=Sippy 154417260 0 IN IP4 91.121.101.126
    s=SIP Call
    t=0 0
    m=audio 54274 RTP/AVP 18 8 0 98
    c=IN IP4 91.121.75.124
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:98 telephone-event/8000
    a=fmtp:98 0-16
    ***************************** SIP message buffer end ******************************
    
    18:38:31 Try to send SIP message # (13/05/2008 16:38:31:462 GMT) # UDP # 418 bytes # from: 192.168.1.10:5060 # to: 91.121.101.126:5060
    
    ***************************** SIP message buffer start *****************************
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 91.121.101.126;branch=z9hG4bKccc1.022038c2dd4cb31a9b016fbb0c6eb9d9.0
    Via: SIP/2.0/UDP 91.121.101.126:5061;branch=z9hG4bK7584bfd86b96500312c96c574f060abe;rport=5061
    To: <sip:41225500341@91.121.101.126>
    From: <sip:+41764400351@91.121.101.126>;tag=7e82800a29a780c91375fb1c6b984d10
    CSeq: 200 INVITE
    Call-ID: DA24E1F1-204111DD-BF2B857F-4FFC935D@146.188.127.1
    Content-Length: 0
    
    ***************************** SIP message buffer end ******************************
    
    18:38:31 TLayer::MsgToTU # Msg type: 51 # TID: 2757003 # DID: 0
    18:38:31 UACore::TLReqMsgProc # Got INVITE SIP request
    18:38:31 UACore::TLReqMsgProc # Receive out of dialog INVITE request with not self Request-URI address
    18:38:31 UACore::RejectRequest # Request INVITE rejected by 403 response
    18:38:31 Try to send SIP message # (13/05/2008 16:38:31:476 GMT) # UDP # 489 bytes # from: 192.168.1.10:5060 # to: 91.121.101.126:5060
    
    ***************************** SIP message buffer start *****************************
    SIP/2.0 403 Non-self Request-URI
    Via: SIP/2.0/UDP 91.121.101.126;branch=z9hG4bKccc1.022038c2dd4cb31a9b016fbb0c6eb9d9.0
    Via: SIP/2.0/UDP 91.121.101.126:5061;branch=z9hG4bK7584bfd86b96500312c96c574f060abe;rport=5061
    To: <sip:41225500341@91.121.101.126>
    From: <sip:+41764400351@91.121.101.126>;tag=7e82800a29a780c91375fb1c6b984d10
    CSeq: 200 INVITE
    Call-ID: DA24E1F1-204111DD-BF2B857F-4FFC935D@146.188.127.1
    Server: Epygi Quadro SIP User Agent/v4.1.52 (QUADRO-2X)
    Content-Length: 0
    
    ***************************** SIP message buffer end ******************************
    
    
    ------------------- Application Log Started At 2008/05/13 18:38:31 -------------------
    
    18:38:31 Receive SIP message # (13/05/2008 16:38:31:544 GMT) # UDP # 404 bytes # from: 91.121.101.126:5060 # to: 192.168.1.10:5060
    
    ***************************** SIP message buffer start *****************************
    ACK sip:41225500341@213.221.210.139:5060 SIP/2.0
    Via: SIP/2.0/UDP 91.121.101.126;branch=z9hG4bKccc1.022038c2dd4cb31a9b016fbb0c6eb9d9.0
    From: <sip:+41764400351@91.121.101.126>;tag=7e82800a29a780c91375fb1c6b984d10
    Call-ID: DA24E1F1-204111DD-BF2B857F-4FFC935D@146.188.127.1
    To: <sip:41225500341@91.121.101.126>
    CSeq: 200 ACK
    User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
    Content-Length: 0
    
    ***************************** SIP message buffer end ******************************

  2. #2

    Default

    Umm.. I just did some testing since I have various providers.. and it seems my PBX acts the same way for a variety of providers now:

    Phonestar.ch
    Dynamic-phone.ch
    sipnumber.com
    switzernet.com

    I have but two accounts where I'm still reachable
    Netvoip.ch
    sipgate.de

    @edit: Hmm.. when I saw that there was a new firmware I just had to upgrade and after the upgrade plus the additional reboot to reload my config and booting all the phones things are back to normal. But if anybody has an idea how this could've happened I'd still like to know as I need to prevent racking up such costs (the overflow is a safety mechanism so that we're always reachable even if the internet link goes down.. ).
    Last edited by ssteiner; 05-13-2008 at 12:43 PM.

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