[test-results] [Bamboo] Asterisk Testing > Certified Asterisk 1.8.11 Branch > #20 was SUCCESSFUL (with 182 tests). Change made by qwell.

[Bamboo]

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Asterisk Testing > Certified Asterisk 1.8.11 Branch > #20 was successful.
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Code has been updated by qwell.
182 tests in total.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKCERTIFIED1811-20/


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Code Changes
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qwell (369839):

>Re-merge changes that were reverted.
>
>------------------------------------------------------------------------
>r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) | 7 lines
>
>Add support for folders in MixMonitor 'm' option.  Backport manager actions.
>
>The manager actions are needed, so MixMonitor can be executed on existing
>channels.
>
>(issue DPMA-68)
>
>------------------------------------------------------------------------
>r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) | 6 lines
>
>Remove folder_dir from voicemail snapshots API.
>
>It was both unused (except in tests, where it was fudged) and unnecessary.
>
>(closes issue AST-842)
>
>------------------------------------------------------------------------
>r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May 2012) | 21 lines
>
>Add "send to voicemail" Digium phone functionality to Asterisk.
>
>This change accommodates two methods by which calls can be directed to
>a user's voicemail.
>
>* Incoming calls can be redirected to any user's voicemail.
>* Established calls can be blind transferred to any user's voicemail.
>
>Digium phones indicate the desire to direct a call to voicemail by using
>a Diversion header with a reason parameter of "send_to_vm".
>
>This patch adds the "send_to_vm" reason as a valid redirecting reason. In
>addition, chan_sip.c has been modified to update redirecting information
>on the transferred channel by reading a Diversion header on a REFER request.
>
>(closes issue AST-871)
>Reported by Malcolm Davenport
>
>Review: https://reviewboard.asterisk.org/r/1925
>
>------------------------------------------------------------------------
>r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012) | 18 lines
>
>Fix deadlock in SIP transfers that involve a REFER request
>
>In r367163, "send to voicemail" functionality was added to the SIP channel
>driver.  This required updating the party redirecting information for the
>channel based on the headers provided in the REFER request.  When the
>redirecting party information is updated on the channel, a call to
>ast_indicate_data occurs.  Because handle_request_refer still had the sip_pvt
>locked, a deadlock could occur between the pbx_thread and the do_monitor thread
>servicing the REFER request.
>
>This patch preserves the proper locking order between the channel and the
>sip_pvt by ensuring that the sip_pvt is unlocked prior to updating the party
>redirecting information on the channel.
>
>(closes issue AST-903)
>Reported by: Matt Jordan
>patches:
>  jira_ast_903_trunk.patch by rmudgett (license 5621)
>
>------------------------------------------------------------------------
>r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) | 11 lines
>
>Remove global symbol requirement from app_voicemail.
>
>This uses the existing "function installation" stuff that already existed for
>other functions, like getting message counts.
>
>(closes issue AST-807)
>(issue AST-901)
>(issue AST-908)
>
>Review: https://reviewboard.asterisk.org/r/1965/
>
>------------------------------------------------------------------------
>r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) | 8 lines
>
>These functions that were moved need to be static.
>
>Also wrap test functions in a #ifdef.
>
>(issue AST-807)
>(issue AST-901)
>(issue AST-908)
>
>------------------------------------------------------------------------
>r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) | 6 lines
>
>Remove some symbol exports that got missed in the removal of global symbols.
>
>(issue AST-807)
>(issue AST-901)
>(issue AST-908)
>
>------------------------------------------------------------------------
>r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) | 2 lines
>
>Fix voicemail API tests by using the correct argument order for create/destroy.
>
>------------------------------------------------------------------------
>

qwell (369840):

>Remove file that should no longer exist.



--------------
Tests
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Fixed Tests (9)
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record temp
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record unavail
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record name
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding
   - AsteriskTestSuite: S/apps/voicemail/check voicemail new user
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail priority
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail nominal
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record busy
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding auto urgent

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