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

[Tilghman Lesher]

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/trunk/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/591/#comment3852>

    Just so as to leave no doubt, this is the line of the actual code fix.


- Tilghman


On 2010-03-25 23:34:39, Tilghman Lesher wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/591/
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> (Updated 2010-03-25 23:34:39)
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> Review request for Asterisk Developers.
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> 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.
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> The fix itself is a one-liner, but the additional changes were needed in order to build a regression test for this scenario.
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> This addresses bug 16774.
>     https://issues.asterisk.org/view.php?id=16774
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> 
> 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 
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> Diff: https://reviewboard.asterisk.org/r/591/diff
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> 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.
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> Thanks,
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> Tilghman
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