[test-results] [Bamboo] Asterisk - 10 > Ubuntu Lucid (10.04) > #153 has FAILED (4 tests failed, 1 failures were new). Change made by rmudgett and jrose.

[Bamboo]

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Asterisk - 10 > Ubuntu Lucid (10.04) > #153 failed.
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Code has been updated by rmudgett, jrose.
4/170 tests failed, 1 failure was new.

http://bamboo.asterisk.org/browse/AST10-LUCID-153/


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Failing Jobs
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  - amd64 (Default Stage): 4 of 170 tests failed.


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Code Changes
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jrose (336717):

>Merged revisions 336716 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/1.8
>
>........
>  r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | 7 lines
>  
>  Document applications that play audio and do not answer unanswered calls.
>  
>  This patch is part of an effort to document early media and its usage. If you are
>  interested in contributing to this documentation effort, there are probably other
>  applications worth documenting as well as an Asterisk wiki article at
>  https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
>........
>

rmudgett (336659):

>Merged revisions 336658 via svnmerge from 
>https://origsvn.digium.com/svn/asterisk/branches/1.8
>
>........
>  r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
>  
>  Made Dial d and H options no longer immediately auto-answer the calling leg.
>  
>  The Dial d and H options break DTMF attended transfer atxferdropcall
>  option.
>  
>  1) Party A calls party B.
>  2) Party B does a DTMF attended transfer to Party C.
>  
>  If the dialplan uses the Dial d or H options to call Party C then the Dial
>  application answers the call immediately before initiating the call leg to
>  Party C.  The premature answer causes the transfer code to not invoke the
>  atxferdropcall=no behavior for a blonde transfer since Party C has
>  "answered".  The transfer code thinks that Party B has "consulted" with
>  Party C when Party B hangs up and completes the transfer to Party A.
>  Party A now hears ringback until Party C actually answers.
>  
>  ASTERISK-13294 Dial d option.
>  ASTERISK-11067 Dial H option to disconnect before answer.
>  
>  The referenced issues made Dial answer with the d and H options because
>  many SIP and ISDN phones cannot send DTMF before the call is connected.
>  
>  * Made require the dialplan to control when or if the call needs to be
>  answered to use the Dial application d and H options.  (The call is no
>  longer surprise answered when using the Dial d or H options.)
>  
>  Review: https://reviewboard.asterisk.org/r/1381/
>  
>  JIRA AST-623
>  JIRA AST-666
>........
>


--------------
Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/fax/gateway timeout1
Existing Test Failures (3)
   - AsteriskTestSuite: S/fax/gateway native t38
   - AsteriskTestSuite: S/fax/gateway native t38 ced
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options change password

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