[test-results] [Bamboo] Asterisk Testing > Asterisk 11 Branch > #100 has FAILED (2 tests failed). Change made by Matthew Jordan, rmudgett and jrose.

[Bamboo]

-----------------------------------------------------------------------
Asterisk Testing > Asterisk 11 Branch > #100 failed.
-----------------------------------------------------------------------
Code has been updated by Matthew Jordan, rmudgett, jrose.
2/286 tests failed.

http://bamboo.asterisk.org/browse/TESTING-AST11BRANCH-100/


--------------
Failing Jobs
--------------
  - Asterisk CentOS 6 64-Bit (CentOS 6): 2 of 286 tests failed.



--------------
Code Changes
--------------
Matthew Jordan (373508):

>Revert change to res_rtp_asterisk committed in r373236 (1.8)
>
>The change committed in r373236 attempted to account for endpoints that
>increased their RTP timestamp in DTMF end of event re-transmissions.  This
>change attempted to make Asterisk continue to work with endpoints that
>failed to follow the RFC while maintaining the fix that allowed for out of
>order DTMF to be handled.  Unfortunately, there is no free lunch, and this
>patch broke any system that sent DTMF immediately after an RTP session was
>established or when an SSRC is updated.  As such, that patch is being
>reverted for the previous behavior.
>
>Endpoints that erroneously increase the RTP timestamp in DTMF end of event
>packets will not work properly with Asterisk.
>
>(issue ASTERISK-20424)
>........
>
>Merged revisions 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 373505 from http://svn.asterisk.org/svn/asterisk/branches/10
>

jrose (373470):

>func_audiohookinherit: Document some missed sources.
>
>This patch also mentions that AUDIOHOOK_INHERIT can be used to
>transfer MixMonitor audiohooks. There is also wiki that addresses
>audiohooks and the use of AUDIOHOOK_INHERIT at the following link:
>https://wiki.asterisk.org/wiki/display/AST/Audiohooks
>
>(closes issue ASTERISK-18220)
>Reported by: Ishfaq Malik
>........
>
>Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 373468 from http://svn.asterisk.org/svn/asterisk/branches/10
>

rmudgett (373502):

>Be consistent, send From: "Anonymous" <sip:anonymous at anonymous.invalid>
>
>When setting CALLERID(pres)=unavailable in the dialplan, the From header
>in the SIP message contains "Anonymous" <sip:Anonymous at anonymous.invalid>.
>For consistency, Asterisk should use a lowercase a in the userpart of the
>URI.
>
>* Make the From header use a lowercase A in the userpart of the anonymous
>URI.
>
>(closes issue ASTERISK-19838)
>Reported by: Antti Yrjola
>Patches:
>      chan_sip_patch_ASTERISK-19838.patch (license #6383) patch uploaded by Antti Yrjola
>........
>
>Merged revisions 373500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 373501 from http://svn.asterisk.org/svn/asterisk/branches/10
>



--------------
Tests
--------------
New Test Failures (2)
   - AsteriskTestSuite: S/channels/ s i p/sip attended transfer tcp
   - AsteriskTestSuite: S/channels/ s i p/ s d p attribute passthrough
Fixed Tests (6)
   - AsteriskTestSuite: S/apps/directory operator exit
   - AsteriskTestSuite: S/apps/voicemail/authenticate extensions
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail contexts
   - AsteriskTestSuite: S/apps/directory attendant exit
   - AsteriskTestSuite: S/apps/voicemail/authenticate nominal
   - AsteriskTestSuite: S/apps/directory context operator exit

--
This message is automatically generated by Atlassian Bamboo
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/test-results/attachments/20120926/c7d094ff/attachment-0001.htm>