Files
asterisk/channels
Mark Michelson 6af217578e Merged revisions 189097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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  r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines
  
  Prevent a crash when SIP blonde transferring an unbridged call.
  
  If one attempts to use the attended transfer button on a SIP phone
  to transfer an unbridged call (such as a call to an IVR) but hangs
  up while the target of the transfer is still ringing, we need to not
  crash.
  
  The problem was that ast_hangup was called from outside the channel
  thread.
  
  AST-211
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@189103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-17 20:21:26 +00:00
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2008-05-22 17:50:40 +00:00