[test-results] [Bamboo] Asterisk - Trunk > FreeBSD 8.1 > #406 has FAILED (62 tests failed, no failures were new). Change made by 4 authors.

[Bamboo]

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Asterisk - Trunk > FreeBSD 8.1 > #406 failed.
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This build occurred because it is a dependant of ASTTRUNK-LUCID-1305.
62/178 tests failed, no failures were new.

http://bamboo.asterisk.org/browse/ASTTRUNK-FREEBSD81-406/


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Failing Jobs
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  - i386 (Default Stage): 62 of 178 tests failed.


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Code Changes
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seanbright (349046):

>Merged revisions 349045 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/10
>
>................
>  r349045 | seanbright | 2011-12-23 12:32:33 -0500 (Fri, 23 Dec 2011) | 25 lines
>  
>  Merged revisions 349044 via svnmerge from 
>  https://origsvn.digium.com/svn/asterisk/branches/1.8
>  
>  ........
>    r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines
>    
>    In ChanSpy, don't create audiohooks that will never be used.
>    
>    When ChanSpy is initialized it creates and attaches 3 audiohooks:
>    
>      1) Read audio off of the channel that we are spying on
>      2) Write audio to the channel that we are spying on
>      3) Write audio to the channel that is bridged to the channel that we are
>         spying on.
>    
>    The first is always necessary, but the others are used only when specific
>    options are passed to the ChanSpy application (B, d, w, and W to be specific).
>    
>    When those flags are not passed, neither of those audiohooks are ever sent
>    frames, but we still try to process the hooks for each voice frame that we
>    recieve on the channel.
>    
>    So in short - only create and attach audiohooks that we actually need.
>  ........
>................
>

kmoore (348994):

>Fix missing doc tags found while fixing ASTERISK-18689
>
>Add missing <variable></variable> tags in app_dial documentation.
>........
>
>Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 348993 from http://svn.asterisk.org/svn/asterisk/branches/10
>

Matthew Jordan (349106):

>Allow overriding of IMAP server settings on a user by user basis
>
>This patch allows the imapserver, imapport, and imapflags settings to be
>overridden for any voicemail user.  It also documents the settings in
>the sample voicemail.conf file, and updates the voicemail schema to
>allow storage of those columns.
>
>(closes issue ASTERISK-16489)
>Reporter: Hubert Mickael
>Tested by: Matt Jordan
>
>Review: https://reviewboard.asterisk.org/r/1614/
>
>


--------------
Tests
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Existing Test Failures (62)
   - 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/channels/ s i p/options
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding
   - AsteriskTestSuite: S/apps/voicemail/check voicemail dialout
   - AsteriskTestSuite: S/channels/ s i p/nat supertest
   - 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/channels/ s i p/sip register domain acl
   - AsteriskTestSuite: S/fax/gateway t38 g711
   - AsteriskTestSuite: S/fax/gateway g711 t38
   - AsteriskTestSuite: S/channels/ s i p/realtime nosipregs
   - AsteriskTestSuite: S/fax/gateway native t38 ced
   - AsteriskTestSuite: S/udptl
   - AsteriskTestSuite: S/apps/confbridge/confbridge nominal
   - AsteriskTestSuite: S/apps/directory context operator exit
   - AsteriskTestSuite: S/fax/gateway timeout5

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