[asterisk-dev] [Code Review] Fix chan_sip websocket handling.
Pedro Kiefer
reviewboard at asterisk.org
Wed Nov 28 11:03:01 CST 2012
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2219/
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Review request for Asterisk Developers and Joshua Colp.
Summary
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Websocket by default doesn't return an ast_str for the payload received. When converting it to an ast_str on chan_sip the last character was being omitted, because ast_str functions expects that the given length includes the trailing 0x00. payload_len only has the actual string length without counting the trailing zero. For most cases this passed unnoticed as most of SIP messages ends with \r\n.
This addresses bug ASTERISK-20745.
https://issues.asterisk.org/jira/browse/ASTERISK-20745
Diffs
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/trunk/channels/chan_sip.c 376616
Diff: https://reviewboard.asterisk.org/r/2219/diff
Testing
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Tested, everything works as supposed.
Thanks,
Pedro
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