[asterisk-bugs] [Asterisk 0014618]: [patch] sip channel freezed in ChanSpy() app
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Nov 20 11:43:44 CST 2009
A NOTE has been added to this issue.
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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
Target Version: 1.6.0.19
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: 2009-11-20 11:43 CST
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Summary: [patch] 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 ;)
======================================================================
----------------------------------------------------------------------
(0114051) svnbot (reporter) - 2009-11-20 11:43
https://issues.asterisk.org/view.php?id=14618#c114051
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Repository: asterisk
Revision: 230587
_U branches/1.6.0/
U branches/1.6.0/include/asterisk/audiohook.h
U branches/1.6.0/main/audiohook.c
------------------------------------------------------------------------
r230587 | dvossel | 2009-11-20 11:43:43 -0600 (Fri, 20 Nov 2009) | 13
lines
Merged revisions 230583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6
lines
audiohook signal trigger on every status change
(issue https://issues.asterisk.org/view.php?id=14618)
Review: https://reviewboard.asterisk.org/r/434/
........
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http://svn.digium.com/view/asterisk?view=rev&revision=230587
Issue History
Date Modified Username Field Change
======================================================================
2009-11-20 11:43 svnbot Checkin
2009-11-20 11:43 svnbot Note Added: 0114051
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