[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
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Project:                    Asterisk
Issue ID:                   13538
Category:                   Applications/app_mixmonitor
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     assigned
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:              
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Date Submitted:             2008-09-23 00:02 CDT
Last Modified:              2008-12-09 15:29 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.
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Relationships       ID      Summary
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related to          0007717 MixMonitor stops after attended call tr...
has duplicate       0013554 Mixmonitor doens't record call after at...
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 (0096037) putnopvut (administrator) - 2008-12-09 15:29
 http://bugs.digium.com/view.php?id=13538#c96037 
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For those following this issue, I have an update. I have started a branch
to correct this issue. It may be checked out from the following location
using subversion:

http://svn.digium.com/svn/asterisk/team/mmichelson/issue13538

If you install this and start Asterisk, you can run "core show function
AUDIOHOOK_INHERIT" and see how to use the AUDIOHOOK_INHERIT function to
allow for MixMonitor to stay alive during a transfer. The support is not
100% complete yet because the functionality described to allow for the
final leg of a transfer to block the inheritance of an audiohook is not yet
working.

Caution: This branch is based off of Asterisk trunk, not Asterisk 1.4. It
is not a good idea to run this in production, and upgrading to it from 1.4
may require work to update your dialplan as well as any external Asterisk
modules you may be using.

I will eventually make a patch for 1.4 and post it here, but I want to go
through the process of testing and review prior to doing that. Thanks for
your patience 

Issue History 
Date Modified    Username       Field                    Change               
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2008-12-09 15:29 putnopvut      Note Added: 0096037                          
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