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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3987/
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Review request for Asterisk Developers.
Bugs: ASTERISK-24211
https://issues.asterisk.org/jira/browse/ASTERISK-24211
Repository: Asterisk
Description
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This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge could
cause a bouncing native bridge. In the case of the dial_LS_options test, this
was a remote RTP bridge which caused the audio path to continually cycle
between Asterisk and the remote endpoints generating a large number of SIP
messages and delaying the test long enough to cause it to fail (checking timing
was part of the test). The root cause was that the code to decide whether to
use native bridging was expecting a time-remaining value of 0 to be the default
instead of the actual default value of -1. A value of 0 or negative numbers
could also be generated by preceding code in some circumstances. Both issues
are addressed in this patch.
Diffs
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branches/1.8/main/channel.c 422850
Diff: https://reviewboard.asterisk.org/r/3987/diff/
Testing
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Verified that the test (11-only) operated correctly with this patch.
Thanks,
opticron
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