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

Fri Aug 21 03:49:22 CDT 2009

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=14618 
====================================================================== 
Reported By:                caspy
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14618
Category:                   Applications/app_chanspy
Reproducibility:            sometimes
Severity:                   minor
Priority:                   normal
Status:                     feedback
Target Version:             1.6.0.14
Asterisk Version:           1.6.0.6 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-03-06 14:35 CST
Last Modified:              2009-08-21 03:49 CDT
====================================================================== 
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 ;)

====================================================================== 

---------------------------------------------------------------------- 
 (0109446) caspy (reporter) - 2009-08-21 03:49
 https://issues.asterisk.org/view.php?id=14618#c109446 
---------------------------------------------------------------------- 
one more interesting fact:
as i said early in comment 0102005, - i saw such freeze on a DAHDI
channel. so, rtptimeout can be not a solution. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-08-21 03:49 caspy          Note Added: 0109446                          
======================================================================