[asterisk-bugs] [Asterisk 0013957]: SIP Channels Hang - Last Message: Rx BYE - Need Destroy: 2
Asterisk Bug Tracker
noreply at bugs.digium.com
Sun Nov 23 18:38:38 CST 2008
The following issue has been SUBMITTED.
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http://bugs.digium.com/view.php?id=13957
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Reported By: geoff2010
Assigned To:
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Project: Asterisk
Issue ID: 13957
Category: Channels/chan_sip/General
Reproducibility: random
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.21.2
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: 2008-11-23 18:38 CST
Last Modified: 2008-11-23 18:38 CST
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Summary: SIP Channels Hang - Last Message: Rx BYE - Need
Destroy: 2
Description:
I have a bunch of servers running 1.4.21.2. They are all, on occassion,
getting stuck SIP channels. Nothing shows up in "core show channels", but
the lingering culprits continue to show up in "sip show channels" until a
restart of asterisk. The thing they all have in common is that the last
message received was a BYE, and they all have their "Need Destroy" flag set
to 2. Here are some basic outputs, nothing is crashing so I have no core
dumps, it's a production system so I don't have SIP debugging enabled. It
seems to only happen in roughly 1 in 5000 calls (estimate)
atl-asterisk5*CLI> core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
-------------------------------------------------------------------------
-------------------------------------------------------------------------
atl-asterisk5*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
Hold Last Message
216.82.XXX.XXX +17066XXX 6b25419936b 00102/00001 0x0 (nothing) No
(d) Rx: BYE
216.82.XXX.XXX +14042XXX 3955ba5d7c2 00102/00001 0x0 (nothing) No
(d) Rx: BYE
2 active SIP channels
-------------------------------------------------------------------------
-------------------------------------------------------------------------
atl-asterisk5*CLI> sip show channel 6b25419936b
atl-asterisk5*CLI>
* SIP Call5*CLI>
Curr. trans. direction: Outgoing
Call-ID: 6b25419936bd5cac4b4dbe6568042787 at blah.com
Owner channel ID: <none>
Our Codec Capability: 260
Non-Codec Capability (DTMF): 1
Their Codec Capability: 256
Joint Codec Capability: 256
Format: 0x0 (nothing)
MaxCallBR: 384 kbps
Theoretical Address: 216.82.XXX.XXX:5060
Received Address: 216.82.XXX.XXX:5060
SIP Transfer mode: open
NAT Support: Always
Audio IP: 67.220.101.185 (local)
Our Tag: as1f608769
Their Tag: VPST506071629460
SIP User agent:
Username: +1706687XXXX
Peername: bw_g729
Original uri: sip:+170668XXXX at 216.82.XXX.XXX
Need Destroy: 2
Last Message: Rx: BYE
Promiscuous Redir: No
Route: N/A
DTMF Mode: rfc2833
SIP Options: (none)
If there is anything else I can provide, please let me know.
Thanks,
Geoff
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Issue History
Date Modified Username Field Change
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2008-11-23 18:38 geoff2010 New Issue
2008-11-23 18:38 geoff2010 Asterisk Version => 1.4.21.2
2008-11-23 18:38 geoff2010 SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
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