[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #261 has FAILED (3 tests failed, 2 failures were new). Change made by root.

[Bamboo]

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Asterisk - Team Branches > Pimp My SIP > #261 failed.
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Code has been updated by root.
1/2 jobs failed, with 3 failing tests, 2 failures were new.

http://bamboo.asterisk.org/browse/ASTTEAM-PIMPMYSIP-261/


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Failing Jobs
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  - Asterisk 1.8 CentOS 6 32-Bit (CentOS 6): 3 of 271 tests failed.



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Code Changes
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root (385640):

>Multiple revisions 385635,385638
>
>........
>  r385635 | mjordan | 2013-04-13 21:35:04 -0500 (Sat, 13 Apr 2013) | 23 lines
>  
>  Don't attempt to create a voice frame on a read error
>  
>  Prior to this patch, a read error in snd_pcm_readi would still be treated as a
>  nominal result when constructing a voice frame from the expected data. Since
>  the value returned is negative, as opposed to the number of samples read,
>  this could result in a crash. With this patch, we now return a null frame
>  when a read error is detected.
>  
>  Note that the patch on ASTERISK-21329 was modified slightly for this commit,
>  in that we bail immediately on detecting the read error, rather than bypassing
>  the construction of the voice frame.
>  
>  (closes issue ASTERISK-21329)
>  Reported by: Keiichiro Kawasaki
>  patches:
>    chan_alsa.diff uploaded by kawasaki (License 6489)
>  ........
>  
>  Merged revisions 385633 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 385634 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r385638 | mjordan | 2013-04-13 22:01:33 -0500 (Sat, 13 Apr 2013) | 19 lines
>  
>  Calculate the timestamp for outbound RTP if we don't have timing information
>  
>  This patch calculates the timestamp for outbound RTP when we don't have timing
>  information. This uses the same approach in res_rtp_asterisk. Thanks to both
>  Pietro and Tzafrir for providing patches.
>  
>  (closes issue ASTERISK-19883)
>  Reported by: Giacomo Trovato
>  Tested by: Pietro Bertera, Tzafrir Cohen
>  patches:
>    rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
>    rtp-timestamp.patch uploaded by pbertera (License 5943)
>  ........
>  
>  Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 385637 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>
>Merged revisions 385635,385638 from file:///srv/subversion/repos/asterisk/trunk
>



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Tests
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New Test Failures (2)
   - AsteriskTestSuite: S/channels/gulp/basic calls/incoming/nominal/authed/userpass/ident by user
   - AsteriskUnitTests: /apps/app voicemail/test voicemail notify endl
Existing Test Failures (1)
   - AsteriskTestSuite: S/channels/gulp/incoming calls without auth

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