Details

    • Type: Bug Bug
    • Status: Closed
    • Severity: Major Major
    • Resolution: Fixed
    • Affects Version/s: None
    • Target Release Version/s: None
    • Component/s: Core/General
    • Labels:
      None
    • Mantis ID:
      14431
    • Regression:
      No

      Description

      Our provider says the CANCEL request sends from our asterisk server has always got a different branch parameter value than the branch id sends with the initial request.
      Which creates a major issue that the other party's phone keeps ringing even after the caller hangup the line.

      From RFC3261
      Section 8.1.1.7

      "The branch parameter value MUST be unique across space and time for
      all requests sent by the UA. The exceptions to this rule are CANCEL
      and ACK for non-2xx responses. As discussed below, a CANCEL request
      will have the same value of the branch parameter as the request it
      cancels. As discussed in Section 17.1.1.3, an ACK for a non-2xx
      response will also have the same branch ID as the INVITE whose
      response it acknowledges."

                • ADDITIONAL INFORMATION ******

      U xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
      INVITE sip:0044XXXXXXXXX@sip.myprovider.com SIP/2.0.
      Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK69c66c6e;rport.
      From: "xxxxxxxx" <sip:XXXXXXXXXX@xxx.xxx.xxx.xxx>;tag=as162b95c0.
      To: <sip:0044XXXXXXXXX@sip.myprovider.com>.
      Contact: <sip:XXXXXXXXXX@xxx.xxx.xxx.xxx>.
      Call-ID: 03f854571d10a93606b7cbef3e1d29f7@xxx.xxx.xxx.xxx.
      CSeq: 103 INVITE.
      User-Agent: Asterisk PBX.
      Max-Forwards: 70.
      Proxy-Authorization: Digest username="USERNAME", realm="MY REALM", algorithm=MD5, uri="sip:0044XXXXXXXXX@sip.myprovider.com", nonce="498e3a65000000dcf637040f7ac0b7472fb5cd88cc86bc85", response="7c7c44010293c91a2da6b4b4483c4c6b", qop=auth, cnonce="0569b00d", nc=00000001.
      Date: Sun, 08 Feb 2009 02:03:16 GMT.
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
      Supported: replaces.
      Content-Type: application/sdp.
      Content-Length: 309.
      .

      U xxx.xxx.xxx.xxx:5060 -> yyy.yyy.yyy.yyy:5060
      CANCEL sip:0044XXXXXXXXX@sip.myprovider.com SIP/2.0.
      Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK01ade983;rport.
      From: "XXXXXXXX" <sip:XXXXXXXXX@xxx.xxx.xxx.xxx>;tag=as162b95c0.
      To: <sip:0044XXXXXXXXXX@sip.myprovider.com>.
      Call-ID: 03f854571d10a93606b7cbef3e1d29f7@xxx.xxx.xxx.xxx.
      CSeq: 103 CANCEL.
      User-Agent: Asterisk PBX.
      Max-Forwards: 70.
      Content-Length: 0.

        Activity

        Hide
        Joshua Colp added a comment -

        This has already been fixed by oej in revision 171527.

        Show
        Joshua Colp added a comment - This has already been fixed by oej in revision 171527.

          People

          • Votes:
            0 Vote for this issue
            Watchers:
            0 Start watching this issue

            Dates

            • Created:
              Updated:
              Resolved:

              Development