[test-results] [Bamboo] Asterisk - Trunk > FreeBSD 8.1 > #403 has FAILED (61 tests failed, 8 failures were new). Change made by 16 authors.
    Bamboo 
    bamboo at asterisk.org
       
    Tue Dec 20 23:49:10 CST 2011
    
    
  
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Asterisk - Trunk > FreeBSD 8.1 > #403 failed.
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This build occurred because it is a dependant of ASTTRUNK-LUCID-1292.
61/172 tests failed, 8 failures were new.
http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-403/
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Failing Jobs
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  - i386 (Default Stage): 61 of 172 tests failed.
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Code Changes
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jrose (348156):
>Document PARKINGSLOT variable in features.conf.sample
>
>(issue ASTERISK-16239)
>........
>
>Merged revisions 348154 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 348155 from http://svn.asterisk.org/svn/asterisk/branches/10
>
jrose (347866):
>Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps.
>
>When a caller sends DTMF while the SayUnixTime application is saying the time, The call
>would jump to the next extension much like it does during Background(). This patch adds
>option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
>allows arguments to sayunixtime to not be used as empty strings in the case of something
>like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).
>
>(closes issue ASTERISK-16675)
>Reported by: jlpedrosa
>Patches:
>	patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
>Review: https://reviewboard.asterisk.org/r/956/
>
rmudgett (348408):
>Fix cut and past error in ast_call_forward().
>
>(issue ASTERISK-18836)
>........
>
>Merged revisions 348401 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 348405 from http://svn.asterisk.org/svn/asterisk/branches/10
>
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Tests
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New Test Failures (8)
   - AsteriskTestSuite: S/channels/ s i p/options
   - AsteriskTestSuite: S/channels/ s i p/nat supertest
   - AsteriskTestSuite: S/channels/ s i p/sip register domain acl
   - AsteriskTestSuite: S/dialplan
   - AsteriskTestSuite: S/channels/ s i p/noload res srtp
   - AsteriskTestSuite: S/queues/queue baseline
   - AsteriskTestSuite: S/queues/set penalty
   - AsteriskTestSuite: S/channels/ s i p/sip tls register
Existing Test Failures (53)
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail priority
   - AsteriskTestSuite: S/apps/voicemail/check voicemail reply
   - AsteriskTestSuite: S/fastagi/connect
   - AsteriskTestSuite: S/apps/voicemail/check voicemail new user
   - AsteriskTestSuite: S/apps/voicemail/func vmcount
   - AsteriskTestSuite: S/fastagi/stream-file
   - AsteriskTestSuite: S/directed pickup
   - AsteriskTestSuite: S/apps/directory operator exit
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding
   - AsteriskTestSuite: S/apps/voicemail/check voicemail dialout
   - AsteriskTestSuite: S/apps/incomplete/sip incomplete
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding auto urgent
   - AsteriskTestSuite: S/apps/voicemail/check voicemail envelope
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options change password
   - AsteriskTestSuite: S/fax/gateway t38 g711
   - AsteriskTestSuite: S/fax/gateway g711 t38
   - AsteriskTestSuite: S/channels/ s i p/realtime nosipregs
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