[asterisk-bugs] [Asterisk 0018837]: [patch] Deadlock with attended transfer of SIP call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18837 
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Reported By:                alecdavis
Assigned To:                alecdavis
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Project:                    Asterisk
Issue ID:                   18837
Category:                   Core/RTP
Reproducibility:            always
Severity:                   crash
Priority:                   normal
Status:                     closed
Asterisk Version:           1.8.2.3 
JIRA:                        
Regression:                 No 
Reviewboard Link:           https://reviewboard.asterisk.org/r/1126/ 
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:             2011-02-17 19:25 CST
Last Modified:              2011-02-25 12:58 CST
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Summary:                    [patch] Deadlock with attended transfer of SIP call
Description: 
3 SIP phones.

A calls B, and B answers on line 1.
B puts A on hold by selecting line2.
B calls C, and C answers.
B initiates transfer of line1 to line2, phone uses REFER.


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Relationships       ID      Summary
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related to          0018491 [patch] fix SIP indicate deadlocks when...
has duplicate       0018468 SIP crash on transfer
has duplicate       0018734 Combination dtmfmode=info, directmedia=...
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 (0132416) svnbot (reporter) - 2011-02-25 12:58
 https://issues.asterisk.org/view.php?id=18837#c132416 
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Repository: asterisk
Revision: 308946

_U  trunk/
U   trunk/channels/chan_sip.c

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r308946 | alecdavis | 2011-02-25 12:58:11 -0600 (Fri, 25 Feb 2011) | 27
lines

Merged revisions 308945 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb 2011) | 21
lines
  
  Fix Deadlock with attended transfer of SIP call
  
  Call path 
    sip_set_rtp_peer (locks chan then pvt)
     transmit_reinvite_with_sdp
      try_suggested_sip_codec
       pbx_builtin_getvar_helper (locks p->owner)
  
  But by the time p->owner lock was attempted, seems as though chan and
p->owner were different.
  
  So in sip_set_rtp_peer, lock pvt first then lock p->owner using
deadlocking methods.
  
  (closes issue https://issues.asterisk.org/view.php?id=18837)
  Reported by: alecdavis
  Patches: 
        bug18837-trunk.diff3.txt uploaded by alecdavis (license 585)
  Tested by: alecdavis, Irontec, ZX81, cmaj
  
  Review: [https://reviewboard.asterisk.org/r/1126/]
........

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http://svn.digium.com/view/asterisk?view=rev&revision=308946 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-25 12:58 svnbot         Checkin                                      
2011-02-25 12:58 svnbot         Note Added: 0132416                          
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