[asterisk-bugs] [Asterisk 0014355]: [patch] Segfault if you transfer a call into a meetme room

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14355 
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Reported By:                sodom
Assigned To:                rmudgett
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Project:                    Asterisk
Issue ID:                   14355
Category:                   Channels/chan_misdn
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.23 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-01-28 08:44 CST
Last Modified:              2009-07-14 12:01 CDT
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Summary:                    [patch] Segfault if you transfer a call into a
meetme room
Description: 
Asterisk crashes with a seg fault if you transfer a call (via ISDN
Transfer) into a meetme room. 
Hold/Retrive between PRI-Zaptel and/or other mISDN channels works.

e.g. 
Phone No: 10 connected to port 3
Phone No: 11 connected to port 4
Meetme: Call No: 771 routes into Meetme-Room 9991 (static)

You connect 11 to 10 (success)
-> then press R for hold on 11 (success), 10 gets MOH
-> dial 771 on 11 (success) you get anouncement of MeetMe
-> hang up 11 to transfer 10 to MeetMe -> (crash, seg fault)
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---------------------------------------------------------------------- 
 (0107720) svnbot (reporter) - 2009-07-14 12:01
 https://issues.asterisk.org/view.php?id=14355#c107720 
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Repository: asterisk
Revision: 206489

_U  trunk/
U   trunk/channels/chan_misdn.c
U   trunk/channels/misdn/isdn_lib.c
U   trunk/channels/misdn/isdn_lib.h

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r206489 | rmudgett | 2009-07-14 12:01:49 -0500 (Tue, 14 Jul 2009) | 35
lines

Merged revisions 206487 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r206487 | rmudgett | 2009-07-14 11:44:47 -0500 (Tue, 14 Jul 2009) | 28
lines
  
  Fixes several call transfer issues with chan_misdn.
  
  *  issue https://issues.asterisk.org/view.php?id=14355 - Crash if attempt to
transfer a call to an application.
  Masquerade the other pair of the four asterisk channels involved in the
  two calls.  The held call already must be a bridged call (not an
  applicaton) or it would have been rejected.
  
  *  issue https://issues.asterisk.org/view.php?id=14692 - Held calls are not
automatically cleared after
transfer.
  Allow the core to initate disconnect of held calls to the ISDN port. 
This
  also fixes a similar case where the party on hold hangs up before being
  transferred or taken off hold.
  
  *  JIRA ABE-1903 - Orphaned held calls left in music-on-hold.
  Do not simply block passing the hangup event on held calls to asterisk
  core.
  
  *  Fixed to allow held calls to be transferred to ringing calls.
  Previously, held calls could only be transferred to connected calls.
  *  Eliminated unused call states to simplify hangup code.
  *  Eliminated most uses of "holded" because it is not a word.
  
  (closes issue https://issues.asterisk.org/view.php?id=14355)
  (closes issue https://issues.asterisk.org/view.php?id=14692)
  Reported by: sodom
  Patches:
        misdn_xfer_v14_r205839.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett
........

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http://svn.digium.com/view/asterisk?view=rev&revision=206489 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-07-14 12:01 svnbot         Checkin                                      
2009-07-14 12:01 svnbot         Note Added: 0107720                          
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