[asterisk-dev] [Code Review] SIP Notify via AMI or CLI leaks SIP PVTs
opticron
reviewboard at asterisk.org
Thu Jul 28 13:15:58 CDT 2011
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1332/
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Review request for Asterisk Developers.
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.
This addresses bug ASTERISK-18091.
https://issues.asterisk.org/jira/browse/ASTERISK-18091
Diffs
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trunk/channels/chan_sip.c 329610
Diff: https://reviewboard.asterisk.org/r/1332/diff
Testing
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Reproduced the leak and made sure the leak no longer occurred with the change.
Thanks,
opticron
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