[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #1349 has FAILED (221 tests failed, 1 failures were new). Change made by rmudgett.
Bamboo
bamboo at asterisk.org
Fri Jun 7 01:23:36 CDT 2013
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Asterisk Testing > Asterisk Trunk > #1349 failed.
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Code has been updated by rmudgett.
1/2 jobs failed, with 221 failing tests, 1 failure was new.
http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-1349/
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Failing Jobs
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- Asterisk CentOS 6 32-Bit (CentOS 6): 221 of 249 tests failed.
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Code Changes
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rmudgett (390804):
>Refactor chan_dahdi/sig_analog/sig_pri and chan_misdn to use the common transfer functions.
>
>(closes issue ASTERISK-21523)
>Reported by: Matt Jordan
>
>(closes issue ASTERISK-21524)
>Reported by: Matt Jordan
>
>Review: https://reviewboard.asterisk.org/r/2600/
>
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Tests
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New Test Failures (1)
- AsteriskTestSuite: S/apps/mixmonitor audiohook inherit
Existing Test Failures (220)
- AsteriskTestSuite: S/apps/voicemail/check voicemail reply
- AsteriskTestSuite: S/channels/ s i p/session timers/uas originate/no minse small se
- AsteriskTestSuite: S/fastagi/stream-file
- AsteriskTestSuite: S/apps/voicemail/func vmcount
- AsteriskTestSuite: S/channels/ s i p/subscribe
- AsteriskTestSuite: S/channels/ s i p/sip blind transfer/callee with reinvite
- AsteriskTestSuite: S/pbx/call-files
- AsteriskTestSuite: S/apps/directory operator exit
- AsteriskTestSuite: S/channels/ s i p/session timers/uas originate/no minse no se
- AsteriskTestSuite: S/apps/queues/queue transfer callee
- AsteriskTestSuite: S/apps/voicemail/check voicemail dialout
- AsteriskTestSuite: S/channels/ s i p/acl call
- AsteriskTestSuite: S/fax/local channel t38 queryoption
- AsteriskTestSuite: S/apps/dial/dial dtmf hangup cancel
- AsteriskTestSuite: S/udptl v6
- AsteriskTestSuite: S/channels/ s i p/session timers/uas originate/small minse large se
- AsteriskTestSuite: S/channels/ s i p/tcpauthlimit
- AsteriskTestSuite: S/hangup/handlers
- AsteriskTestSuite: S/apps/confbridge/confbridge nominal
- AsteriskTestSuite: S/apps/queues/queue one paused no answer
- AsteriskTestSuite: S/channels/ s i p/sip blind transfer/callee refer only
- AsteriskTestSuite: S/apps/page
- AsteriskTestSuite: S/apps/voicemail/play message
- AsteriskTestSuite: S/cdr/console dial sip transfer
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