[asterisk-bugs] [Asterisk 0012994]: Spamming CLI / logs with 'Remote host can't match request BYE to call...'
noreply at bugs.digium.com
noreply at bugs.digium.com
Sat Jul 5 13:23:58 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12994
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Reported By: pabelanger
Assigned To:
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Project: Asterisk
Issue ID: 12994
Category: Channels/chan_sip/General
Reproducibility: random
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.6.0-beta9
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 07-04-2008 11:33 CDT
Last Modified: 07-05-2008 13:23 CDT
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Summary: Spamming CLI / logs with 'Remote host can't match
request BYE to call...'
Description:
Every 6/7 seconds, our cli is getting spammed (see below).
sip show channel 05b6e9a07669fa0c0a4d88a663b4a2bd at 192.168.20.2
---
* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 05b6e9a07669fa0c0a4d88a663b4a2bd at 192.168.20.2
Owner channel ID: <none>
Our Codec Capability: 6
Non-Codec Capability (DTMF): 1
Their Codec Capability: 4
Joint Codec Capability: 4
Format: 0x0 (nothing)
T.38 support No
Video support No
MaxCallBR: 384 kbps
Theoretical Address: 192.168.20.3:5070
Received Address: 192.168.20.4:5070
SIP Transfer mode: open
NAT Support: RFC3581
Audio IP: 192.168.20.2 (local)
Our Tag: as643fb95e
Their Tag: as643fb95e
SIP User agent: RTCC/3.0.0.0
Peername: sv0071iv
Original uri: sip:sv0071iv.internal.xxx.on.ca:5070
Need Destroy: No
Last Message: Tx: BYE
Promiscuous Redir: No
Route:
sip:sv0071iv.internal.xxx.on.ca:5070;transport=Tcp;maddr=192.168.20.3
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
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oej - 07-05-08 13:23
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Well, the packets you've captured doesn't tell me anything. Capture all the
console output, from the setup of the call to the BYE that fails and we can
try to figure out what happens. Right now the only thing I can say based on
your debug is "Yes, the remote host doesn't recognize the call". And that's
no bug in Asterisk...
Set debug level to 5, verbose to 5, enable sip history and dumphistory in
sip.conf and turn on sip debugging on the console. Capture all of consoles
output to file. Upload that file after studying it.
Issue History
Date Modified Username Field Change
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07-05-08 13:23 oej Note Added: 0089782
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