[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #697 has FAILED. Change made by 6 authors.

[Bamboo]

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Asterisk Testing > Asterisk Trunk > #697 failed.
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Code has been updated by Terry Wilson, Mark Michelson, rmudgett, jrose, file, Kinsey Moore.
No failed tests found, a possible compilation error.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-697/


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Failing Jobs
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  - Asterisk CentOS 6 64-Bit (CentOS 6): 122 tests passed.



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Code Changes
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Mark Michelson (373701):

>Allow for redirecting reasons to be set to arbitrary strings.
>
>This allows for the REDIRECTING dialplan function to be
>used to set the reason to any string.
>
>The SIP channel driver has been modified to set the redirecting
>reason string to the value received in a Diversion header. In
>addition, SIP 480 response reason text will set the redirecting
>reason as well.
>
>(closes issue AST-942)
>reported by Malcolm Davenport
>
>(closes issue AST-943)
>reported by Malcolm Davenport
>
>Review: https://reviewboard.asterisk.org/r/2101
>
>

rmudgett (373636):

>Make rebuild GSM, ilbc, or lpc10 codecs if the respective sources change.
>........
>
>Merged revisions 373618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 373633 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 373635 from http://svn.asterisk.org/svn/asterisk/branches/11
>

jrose (373634):

>chan_sip: Set Quality of Service for video rtp instance
>
>(closes issue ASTERISK-20201)
>Reported by: ddkprog
>Patches:
>    chan_sip.c.diff uploaded by ddkprog (license 6008)
>........
>
>Merged revisions 373617 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 373631 from http://svn.asterisk.org/svn/asterisk/branches/10
>........
>
>Merged revisions 373632 from http://svn.asterisk.org/svn/asterisk/branches/11
>



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