[asterisk-dev] [Code Review] Forward port of SIP request queueing to trunk

[Russell Bryant]

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/trunk/channels/chan_sip.c
<http://reviewboard.digium.com/r/123/#comment674>

    You have to check for failure from sched_add() here.  If it fails, you'd have a reference leak.  It's an edge case, but possible ...
    
    Just as a general note, I took a quick look in chan_sip and noticed the same bug in a few other places.  I figured I'd mention it in case someone watching the list felt like working on it.  ;-)



/trunk/channels/chan_sip.c
<http://reviewboard.digium.com/r/123/#comment673>

    (I can't make this comment go away, ignore this ...)


- Russell


On 2009-01-07 18:40:04, Kevin Fleming wrote:
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> (Updated 2009-01-07 18:40:04)
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> Review request for Asterisk Developers.
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> Summary
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> Due to the significant changes in reference count handling in chan_sip.c between 1.4 and trunk, I'd appreciate a few other pairs of eyes (with brains attached) reviewing this port of the request queueing code to ensure that refcounts are properly handled.
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> Diffs
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>   /trunk/channels/chan_sip.c 167620 
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> Diff: http://reviewboard.digium.com/r/123/diff
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> Testing
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> Compile testing only.
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> Thanks,
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> Kevin
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