[asterisk-bugs] [Asterisk 0014618]: sip channel freezed in ChanSpy() app

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=14618 
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Reported By:                caspy
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   14618
Category:                   Applications/app_chanspy
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                       SWP-268 
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:             2009-03-06 14:35 CST
Last Modified:              2009-11-09 15:04 CST
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Summary:                    sip channel freezed in ChanSpy() app
Description: 
i have a channel that freezed in a strange state. which i can't kill.

Scenario: SIP/1234 dialed number (897795678) that do "{Answer();
ChanSpy(SIP/5678,q); }", and after some hours i see this:

SIP/1234 - is sjphone. it's alredy free, do nothing! call ended on client.
it's already even unreachable. but channel still exist:

*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/1234-b587fc50    897795678 at fromoffice Up      ChanSpy(SIP/5678,q)
1 active channel
1 active call

*CLI> core show channel SIP/1234-b587fc50
 -- General --
           Name: SIP/1234-b587fc50
           Type: SIP
       UniqueID: 1236337972.459555
      Caller ID: 1234
 Caller ID Name: User Name
    DNID Digits: 897795678
       Language: ru
          State: Up (6)
          Rings: 0
  NativeFormats: 0x8 (alaw)
    WriteFormat: 0x40 (slin)
     ReadFormat: 0x8 (alaw)
 WriteTranscode: Yes
  ReadTranscode: No
1st File Descriptor: 106
      Frames in: 123606
     Frames out: 79133
 Time to Hangup: 0
   Elapsed Time: 9h5m25s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: fromoffice
      Extension: 897795678
       Priority: 2
     Call Group: 32768
   Pickup Group: 32768
    Application: ChanSpy
           Data: SIP/5678,q
    Blocking in: (Not Blocking)
      Variables:
RTPAUDIOQOS=ssrc=1088103444;themssrc=265647381;lp=0;rxjitter=0.023820;rxcount=123606;txjitter=0.000000;txcount=79133;rlp=0;rtt=0.000000
SIPCALLID=1AB484D1-80BF-4F1E-97DD-F3B9FC49AA27 at 10.x.x.x
SIPDOMAIN=sipproxy.int.domain.tld
SIPURI=sip:1234 at 10.x.x.x:1000

  CDR Variables:
level 1: clid="User Name" <1234>
level 1: src=1234
level 1: dst=897795678
level 1: dcontext=fromoffice
level 1: channel=SIP/1234-b587fc50
level 1: lastapp=ChanSpy
level 1: lastdata=SIP/5678,q
level 1: start=2009-03-06 14:12:52
level 1: answer=2009-03-06 14:12:52
level 1: duration=32724
level 1: billsec=32724
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1236337972.459555

*CLI> sip show channel 1AB484D1-80BF-4F1E-97DD-F3B9FC49AA27
  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                1AB484D1-80BF-4F1E-97DD-F3B9FC49AA27 at 10.x.x.x
  Owner channel ID:       SIP/1234-b587fc50
  Our Codec Capability:   14
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   1038
  Joint Codec Capability:   14
  Format:                 0x8 (alaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.x.x.x:1000
  Received Address:       10.x.x.x:1000
  SIP Transfer mode:      open
  NAT Support:            Always
  Audio IP:               10.y.y.y (local)
  Our Tag:                as6bde6d25
  Their Tag:              10251567111166
  SIP User agent:         SJphone/1.60.289a (SJ Labs)
  Username:               1234
  Peername:               1234
  Original uri:           sip:1234 at 10.x.x.x:1000
  Caller-ID:              1234
  Need Destroy:           No
  Last Message:           Rx: BYE
  Promiscuous Redir:      No
  Route:                  sip:1234 at 10.x.x.x:1000
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive

*CLI> soft hangup SIP/1234-b587fc50
Requested Hangup on channel 'SIP/1234-b587fc50'


'soft hangup' DO NOTHING. channel is still existing.
i did not yet restart my system, so if i can do anything more for
diagnostic - please tell. this is rare situation, so, if i can look smth
else - i should do it till nearest reload ;)

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---------------------------------------------------------------------- 
 (0113456) dvossel (administrator) - 2009-11-09 15:04
 https://issues.asterisk.org/view.php?id=14618#c113456 
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Can you provide more information exactly what version of Asterisk you are
using? Please provide the branch as well as the svn revision number. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-11-09 15:04 dvossel        Note Added: 0113456                          
2009-11-09 15:04 dvossel        Status                   assigned => feedback
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