[Asterisk-Users] "X-Asterisk-HangupCause: Normal Clearing"

[Stephane Ricard]

Hi,

 

I am initiating a SIP call from Asterisk.  After about 10 minutes, I loose
audio in both directions but the call seem to stay up. Can someone please
help me understand what is happening here.  Been struggling on this for a
while now.  This one is preventing me from fully enjoying my Asterisk
installation :-(

 

Here are the 2 last debug items from the console.

 

  <-- SIP read from 62.123.211.31:5060:

    INFO sip:xxvxxxxxx at 69.63.223.12 SIP/2.0

    t: "STEPHANE RICARD" <sip:xxxxx at xxxxxxxx.ca>;tag=3Das1ea35b0b

    f: <sip:xxxxxxxxxx at xxxxxxxx.ca>;tag=3D47270277584177094

    i: 0781f6c1263705940332c9f5720a7108 at xxxxxxxx.ca

    CSeq: 76518 INFO

    v: SIP/2.0/UDP =

    62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5

    Max-Forwards: 18

    x-nt-corr-id: 10b96be5a191c488003ff5201145bd23ad39c0407 at 62.123.211.31

    k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec

    l: 0

 

  Receiving INFO!

    Transmitting (no NAT) to 62.123.211.31:5060:

    SIP/2.0 403 Unauthorized

    Via: SIP/2.0/UDP =

 
62.123.211.31:5060;branch=3Dz9hG4bK566fcdf06c789739901a9d3fdb0cd2f5;recei=

    ved=3D62.123.211.31

    From: <sip:xxxxxxxxxx at xxxxxxxx.ca>;tag=3D47270277584177094

    To: "STEPHANE RICARD" <sip:xxxxx at xxxxxxxx.ca>;tag=3Das1ea35b0b

    Call-ID: 0781f6c1263705940332c9f5720a7108 at xxxxxxxx.ca

    CSeq: 76518 INFO

    User-Agent: Asterisk PBX

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Contact: <sip:xxxxxxx at 69.63.223.12>

    Content-Length: 0

    X-Asterisk-HangupCause: Normal Clearing

 

Thanks in advance.

Stephane

 

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