[asterisk-dev] [Code Review] Fix descriptor (and structure reference) leak when call is hungup before answer

[Tilghman Lesher]

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https://reviewboard.asterisk.org/r/591/
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Review request for Asterisk Developers.


Summary
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In trunk and current 1.6.x releases, if a SIP channel is hungup prior to being answered, then a reference leak results, which causes RTP ports not to be freed.

The fix itself is a one-liner, but the additional changes were needed in order to build a regression test for this scenario.


This addresses bug 16774.
    https://issues.asterisk.org/view.php?id=16774


Diffs
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  /trunk/channels/chan_sip.c 254405 
  /trunk/channels/sip/dialplan_functions.c 254405 
  /trunk/channels/sip/include/devices.h PRE-CREATION 
  /trunk/channels/sip/include/globals.h 254405 

Diff: https://reviewboard.asterisk.org/r/591/diff


Testing
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Unit test written and tested.  There is one error that running the test still causes, but it's cosmetic and should be able to be tracked down fairly easily.


Thanks,

Tilghman