Stephane Ricard
2006-Jun-03 08:09 UTC
[Asterisk-Users] "X-Asterisk-HangupCause: Normal Clearing"
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@69.63.223.12 SIP/2.0
t: "STEPHANE RICARD"
<sip:xxxxx@xxxxxxxx.ca>;tag=3Das1ea35b0b
f: <sip:xxxxxxxxxx@xxxxxxxx.ca>;tag=3D47270277584177094
i: 0781f6c1263705940332c9f5720a7108@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@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@xxxxxxxx.ca>;tag=3D47270277584177094
To: "STEPHANE RICARD"
<sip:xxxxx@xxxxxxxx.ca>;tag=3Das1ea35b0b
Call-ID: 0781f6c1263705940332c9f5720a7108@xxxxxxxx.ca
CSeq: 76518 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:xxxxxxx@69.63.223.12>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
Thanks in advance.
Stephane
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