[asterisk-bugs] [Asterisk 0013538]: [patch] Recording stops after Transfer when using MixMonitor()

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13538 
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Reported By:                mbit
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   13538
Category:                   Applications/app_mixmonitor
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.21.2 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2008-09-23 00:02 CDT
Last Modified:              2008-12-19 17:07 CST
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Summary:                    [patch] Recording stops after Transfer when using
MixMonitor()
Description: 
When an extension is set to record and the call is transferred to another
extensions which is also recording, the recording stops as soon as the call
is transferred.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0007717 MixMonitor stops after attended call tr...
has duplicate       0013554 Mixmonitor doens't record call after at...
====================================================================== 

---------------------------------------------------------------------- 
 (0096739) svnbot (reporter) - 2008-12-19 17:07
 http://bugs.digium.com/view.php?id=13538#c96739 
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Repository: asterisk
Revision: 166098

_U  branches/1.6.1/
A   branches/1.6.1/funcs/func_audiohookinherit.c
U   branches/1.6.1/include/asterisk/audiohook.h
U   branches/1.6.1/main/audiohook.c
U   branches/1.6.1/main/channel.c

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r166098 | mmichelson | 2008-12-19 17:07:28 -0600 (Fri, 19 Dec 2008) | 44
lines

Merged revisions 166092,166095 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

........
r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28
lines

Adding a new dialplan function AUDIOHOOK_INHERIT

This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel 
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor 
continue recording the call even after the transfer
has completed.

It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.

(closes issue http://bugs.digium.com/view.php?id=13538)
Reported by: mbit
Patches:
      13538.patch uploaded by putnopvut (license 60)
	  Tested by: putnopvut

Review: http://reviewboard.digium.com/r/102/


........
r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5
lines

Remove the verbatim tag from the author line

I could have sworn I already did that before, though...


........

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http://svn.digium.com/view/asterisk?view=rev&revision=166098 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-19 17:07 svnbot         Checkin                                      
2008-12-19 17:07 svnbot         Note Added: 0096739                          
======================================================================