[test-results] [Bamboo] Asterisk - 10 > Ubuntu Lucid (10.04) > #389 has FAILED (5 tests failed, 1 failures were new). Change made by Terry Wilson, rmudgett and tzafrir.
Bamboo
bamboo at asterisk.org
Fri Dec 9 00:08:25 CST 2011
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Asterisk - 10 > Ubuntu Lucid (10.04) > #389 failed.
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Code has been updated by Terry Wilson, rmudgett, tzafrir.
5/182 tests failed, 1 failure was new.
http://bamboo.asterisk.org/browse/AST10-LUCID-389/
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Failing Jobs
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- amd64 (Default Stage): 5 of 182 tests failed.
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Code Changes
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Terry Wilson (347532):
>Don't crash on INFO automon request with no channel
>
>AST-2011-014. When automon was enabled in features.conf, it was possible
>to crash Asterisk by sending an INFO request if no channel had been
>created yet.
>
>(closes issue ASTERISK-18805)
>........
>
>Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
>........
>
>Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>
tzafrir (973):
>menuselect-dummy: fix support_level and more
>
>* Parsing <support_level> requires adding '_' to the regex.
>* Parse attribute in the main tag and treat them the same as sub-tags.
>* ' */' can also terminate MODULEINFO. Be more strict about the end.
> (Make sure we don't get the terminating ' ***/' from the DOCUMENTATION
> in app_macro.c).
>
rmudgett (347600):
>Mark channel running the h exten with the soft-hangup flag.
>
>When a bridge is broken, ast_bridge_call() might execute the h exten on
>the calling channel. However, that channel may not have been the channel
>that broke the bridge by hanging up. The channel executing the h exten
>must be in a hung up state so things like AGI run in the correct mode.
>
>* Make sure ast_bridge_call() marks the channel it is executing the h
>exten on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
>match the pbx.c main dialplan execution loop when it executes the h
>exten.)
>
>(closes issue ASTERISK-18811)
>Reported by: David Hajek
>Patches:
> jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
>Tested by: David Hajek, rmudgett
>........
>
>Merged revisions 347595 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>
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Tests
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New Test Failures (1)
- AsteriskTestSuite: S/queues/queue baseline
Existing Test Failures (4)
- AsteriskTestSuite: S/apps/voicemail/leave voicemail priority
- AsteriskTestSuite: S/channels/ s i p/sip tls register
- AsteriskTestSuite: S/channels/ s i p/sip tls call
- AsteriskTestSuite: S/fax/gateway mix2
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