[asterisk-bugs] [Asterisk 0013957]: SIP Channels Hang - Last Message: Rx BYE - Need Destroy: 2

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
http://bugs.digium.com/view.php?id=13957 
====================================================================== 
Reported By:                geoff2010
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13957
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     feedback
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:              
====================================================================== 
Date Submitted:             2008-11-23 18:38 CST
Last Modified:              2008-11-24 09:18 CST
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0095370) blitzrage (administrator) - 2008-11-24 09:18
 http://bugs.digium.com/view.php?id=13957#c95370 
---------------------------------------------------------------------- 
Per the bug guidelines, we will need sip debugging output:

"6. SIP Problem?
Include debug output! Please include output from "sip debug" if you have a
SIP problem. This seems obvious, but apparently is not. Set debug to 4,
verbose to 4, turn on sip history and dumphistory in sip.conf and capture
all output. A packet trace from ethereal will not tell us what is happening
inside your Asterisk server, so that is not a replacement." 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-24 09:18 blitzrage      Note Added: 0095370                          
2008-11-24 09:18 blitzrage      Status                   new => feedback     
======================================================================