[asterisk-bugs] [Asterisk 0018657]: SIP channel not hung up on BYE

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18657 
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Reported By:                gb_delti
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18657
Category:                   Channels/chan_sip/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.16.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-01-21 07:32 CST
Last Modified:              2011-01-21 13:04 CST
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Summary:                    SIP channel not hung up on BYE
Description: 
I have a SIP peer as a queue member that gets reported as "in use". When I
do a "core show channels", the channel does not show up. When I do "sip
show channels" the channel shows up like this:

10.3.3.234       3044            3869d2204b9d93e   0x100 (g729)     Rx:
BYE

This is the channel info:

 * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                3869d2204b9d93ef
  Owner channel ID:       SIP/3044-00002ba2
  Our Codec Capability:   270
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   268
  Joint Codec Capability:   268
  Format:                 0x100 (g729)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.3.3.234:5060
  Received Address:       10.3.3.234:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               10.3.1.65 (local)
  Our Tag:                as4d7a7e66
  Their Tag:              3fcbfd73a3
  SIP User agent:         Aastra 55i/2.4.1.37
  Username:               3044
  Peername:               3044
  Original uri:           sip:3044 at 10.3.3.234:5060
  Caller-ID:              3044
  Need Destroy:           No
  Last Message:           Rx: BYE
  Promiscuous Redir:      No
  Route:                  sip:3044 at 10.3.3.234:5060;transport=udp
  DTMF Mode:              rfc2833
  SIP Options:            100rel gruu replaces replace timer 
  Session-Timer:          Inactive

I have tried to hang up the channel, but it was not found.
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 (0130879) twilson (administrator) - 2011-01-21 13:04
 https://issues.asterisk.org/view.php?id=18657#c130879 
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If having 'sip set debug' turned on doesn't show BYE retransmissions, then
it isn't the same issue. It looks like in this case Asterisk received a BYE
instead of sent it. There is a transaction timer (sort of) that lasts 64 *
t1timer (by default = 64 * 500ms = 32s) that fires to tear down the sip_pvt
when retransmissions should end. It could be that this is just correct
behavior and that gb_delti just needs to wait the 32 seconds for the
transaction to be torn down. They don't necessarily end when the call ends. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-01-21 13:04 twilson        Note Added: 0130879                          
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