[asterisk-bugs] [Asterisk 0012837]: [patch] Chanspy audio is delayed or lost

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Mar 24 11:51:35 CDT 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=12837 
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Reported By:                pmathews
Assigned To:                mmichelson
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Project:                    Asterisk
Issue ID:                   12837
Category:                   Applications/app_chanspy
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           1.4.20.1 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2008-06-11 16:49 CDT
Last Modified:              2009-03-24 11:51 CDT
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Summary:                    [patch] Chanspy audio is delayed or lost
Description: 
If extension (A) is being monitored via chanspy on extension (C) and (A)
places a call (B) on hold, the audio from the other side of the call (B) is
either delayed or lost to station (C) when the extension being monitored
(A) returns from placing a call on hold. This does not happen if extension
(C) is monitoring extension (B) and extension (A) places extension (B) on
hold.
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Relationships       ID      Summary
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related to          0013745 Recordings out of sync when using chanspy
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---------------------------------------------------------------------- 
 (0102135) renemendoza (reporter) - 2009-03-24 11:51
 http://bugs.digium.com/view.php?id=12837#c102135 
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Hello. This is happening to me on 1.6.0.3, outbound calls from sip
extension to sip trunk, without putting any party ever on hold, delay while
using chanspy is present and eventually will grow to a point where
conversation can not be followed, i see that even patch 13005.tolerance
fixed the problem of out of sync recordings it does not do anything to help
in case of out of sync chan spy. i see that code from patch 13005 is
present in the sources for 1.6.0.3  my sip calls are cristal clear but
monitoring is just not usable as it is 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-03-24 11:51 renemendoza    Note Added: 0102135                          
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