[test-results] [Bamboo] Asterisk Testing > Asterisk 11 Branch > #143 has FAILED (2 tests failed, 1 failures were new). Change made by Mark Michelson, Kinsey Moore and file.

[Bamboo]

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Asterisk Testing > Asterisk 11 Branch > #143 failed.
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Code has been updated by Mark Michelson, file, Kinsey Moore.
2/289 tests failed, 1 failure was new.

http://bamboo.asterisk.org/browse/TESTING-AST11BRANCH-143/


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Failing Jobs
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  - Asterisk CentOS 6 64-Bit (CentOS 6): 2 of 289 tests failed.



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Code Changes
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Mark Michelson (374914):

>Do not use a FILE handle when doing SIP TCP reads.
>
>This is used to solve an issue where a poll on a file
>descriptor does not necessarily correspond to the readiness
>of a FILE handle to be read.
>
>This change makes it so that for TCP connections, we do a
>recv() on the file descriptor instead.
>
>Because TCP does not guarantee that an entire message or even
>just one single message will arrive during a read, a loop has
>been introduced to ensure that we only attempt to handle a
>single message at a time. The tcptls_session_instance structure
>has also had an overflow buffer added to it so that if more
>than one TCP message arrives in one go, there is a place to
>throw the excess.
>
>Huge thanks goes out to Walter Doekes for doing extensive review
>on this change and finding edge cases where code could fail.
>
>(closes issue ASTERISK-20212)
>reported by Phil Ciccone
>
>Review: https://reviewboard.asterisk.org/r/2123
>........
>
>Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 374906 from http://svn.asterisk.org/svn/asterisk/branches/10
>

file (374851):

>Remove code that should not have gotten in.
>
>(issue ASTERISK-20554)

file (374850):

>Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures.
>
>This change removes the requirement for ufrag and pwd in the transport stanza and also
>makes us the controlling agent.
>
>(closes issue ASTERISK-20554)
>Reported by: mmichelson
>



--------------
Tests
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New Test Failures (1)
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record name
Existing Test Failures (1)
   - AsteriskTestSuite: S/channels/ s i p/sip outbound address
Fixed Tests (2)
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record unavail
   - AsteriskTestSuite: S/channels/ s i p/sip attended transfer tcp

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