[asterisk-dev] [Code Review] SIP Notify via AMI or CLI leaks SIP PVTs
David Vossel
reviewboard at asterisk.org
Wed Aug 10 16:58:04 CDT 2011
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https://reviewboard.asterisk.org/r/1332/#review4027
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Ship it!
Nice catch! That was some backwards looking code there.
- David
On July 28, 2011, 1:15 p.m., opticron wrote:
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> https://reviewboard.asterisk.org/r/1332/
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> (Updated July 28, 2011, 1:15 p.m.)
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> Review request for Asterisk Developers.
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> Summary
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> Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. This seems to have occurred when or before transmit_sip_request was replaced by other functions. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed.
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> This addresses bug ASTERISK-18091.
> https://issues.asterisk.org/jira/browse/ASTERISK-18091
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> Diffs
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> trunk/channels/chan_sip.c 329610
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> Diff: https://reviewboard.asterisk.org/r/1332/diff
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> Testing
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> Reproduced the leak and made sure the leak no longer occurred with the change.
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> Thanks,
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> opticron
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