[test-results] [Bamboo] Asterisk - 1.8 > Ubuntu Lucid (10.04) > #782 was SUCCESSFUL (with 146 tests). Change made by 4 authors.

[Bamboo]

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Asterisk - 1.8 > Ubuntu Lucid (10.04) > #782 was successful.
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Code has been updated by Terry Wilson, Russell Bryant, jrose, Tilghman Lesher.
146 tests in total.

http://bamboo.asterisk.org/browse/AST18-LUCID-782/


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Code Changes
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jrose (336716):

>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
>

Tilghman Lesher (336733):

>Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
>
>* Makefile workaround for 10.6 extended to work on 10.7 and later.
>* Now uses the 'weak' symbol for Lion systems, which no longer support
>  'weak_import'
>
>Closes ASTERISK-17612.
>Closes ASTERISK-18213.
>
>Tested by: tilghman, oej.
>

Russell Bryant (336877):

>Fix crashes in ast_rtcp_write().
>
>This patch addresses crashes related to RTCP handling.  The backtraces just
>show a crash in ast_rtcp_write() where it appears that the RTP instance is no
>longer valid.  There is a race condition with scheduled RTCP transmissions and
>the destruction of the RTP instance.  This patch utilizes the fact that
>ast_rtp_instance is a reference counted object and ensures that it will not get
>destroyed while a reference is still around due to scheduled RTCP
>transmissions.
>
>RTCP transmissions are scheduled and executed from the chan_sip scheduler
>context.  This scheduler context is processed in the SIP monitor thread.  The
>destruction of an RTP instance occurs when the associated sip_pvt gets
>destroyed (which happens when the sip_pvt reference count reaches 0).  However,
>the SIP monitor thread is not the only thread that can cause a sip_pvt to get
>destroyed.  The sip_hangup function, executed from a channel thread, also
>decrements the reference count on a sip_pvt and could cause it to get
>destroyed.
>
>While this is being changed anyway, the patch also removes calling
>ast_sched_del() from within the RTCP scheduler callback.  It's not helpful.
>Simply returning 0 prevents the callback from being rescheduled.
>
>(closes issue ASTERISK-18570)
>
>Related issues that look like they are the same problem:
>
>(issue ASTERISK-17560)
>(issue ASTERISK-15406)
>(issue ASTERISK-15257)
>(issue ASTERISK-13334)
>(issue ASTERISK-9977)
>(issue ASTERISK-9716)
>
>Review: https://reviewboard.asterisk.org/r/1444/
>


--------------
Tests
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Fixed Tests (1)
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options change password

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