[test-results] [Bamboo] Asterisk - Team Branches > Pimp My SIP > #345 was SUCCESSFUL (with 434 tests). Change made by Joshua Colp.

Mon May 20 07:59:43 CDT 2013

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Asterisk - Team Branches > Pimp My SIP > #345 was successful.
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Code has been updated by Joshua Colp.
All 2 jobs passed with 434 tests in total.

http://bamboo.asterisk.org/browse/ASTTEAM-PIMPMYSIP-345/


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Code Changes
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Joshua Colp (389196):

>Multiple revisions 388526,388531,388579,388598,388602,388617,388668,388701,388729,388751,388770,388818,388840,388896,388975-388976,389009,389011,389053,389085,389097,389116,389132,389148,389164,389180
>
>........
>  r388526 | jrose | 2013-05-13 14:20:33 -0300 (Mon, 13 May 2013) | 9 lines
>  
>  chan_gulp: Minor readability Improvements to chan_gulp
>  
>  (closes issue ASTERISK-21670)
>  Reported by: Snuffy
>  Review: https://reviewboard.asterisk.org/r/2473/
>  Patches:
>      gulp-coding-guide.diff uploaded by snuffy (license 5024)
>........
>  r388531 | kmoore | 2013-05-13 15:10:22 -0300 (Mon, 13 May 2013) | 15 lines
>  
>  Close libsrtp properly
>  
>  Ensure that libsrtp is shutdown properly when res_srtp is unloaded.
>  
>  (closes issue ASTERISK-21719)
>  Reported by: Corey Farrell
>  Patches:
>      res_srtp-library-shutdown.patch uploaded by Corey Farrell
>  ........
>  
>  Merged revisions 388529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 388530 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388579 | jrose | 2013-05-13 16:29:56 -0300 (Mon, 13 May 2013) | 13 lines
>  
>  pbx: Fix lack of cleanup on macrolock and context_table
>  
>  (closes issue ASTERISK-21723)
>  Reported by: Corey Farrell
>  Patches:
>      core-pbx-cleanup.patch uploaded by Correy Farrell (license 5909)
>  ........
>  
>  Merged revisions 388532 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 388578 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388598 | kmoore | 2013-05-13 17:37:11 -0300 (Mon, 13 May 2013) | 11 lines
>  
>  Revert r388529 for now
>  
>  Adding the cleanup function needs some deeper thought since it
>  apparently doesn't exist for all variants of libsrtp.
>  ........
>  
>  Merged revisions 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 388597 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388602 | elguero | 2013-05-13 18:07:02 -0300 (Mon, 13 May 2013) | 26 lines
>  
>  Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix
>  
>  The prior code committed, r385473, failed to take into consideration that not
>  all outgoing calls will be to a peer.  My fault.
>  
>  This patch does the following:
>  
>  * Check if there is a related peer involved.  If there is, check and set NAT 
>    settings according to the peer's settings.
>  
>  * Fix a problem with realtime peers.  If the global setting has auto_force_rport
>    set and we issued a "sip reload" while a peer is still registered, the peer's
>    flags for NAT are reset to off.  When this happens, we were always setting the
>    contact address of the peer to that of the full contact info that we had.
>  
>  (closes issue ASTERISK-21374)
>  Reported by: jmls
>  Tested by: Michael L. Young
>  Patches:
>     asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)
>  
>  Review: https://reviewboard.asterisk.org/r/2524/
>  ........
>  
>  Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388617 | elguero | 2013-05-13 18:21:03 -0300 (Mon, 13 May 2013) | 18 lines
>  
>  Fix Missing CALL-ID When Logging Through Syslog
>  
>  The CALL-ID (ie [C-00000074]) is missing when logging to syslog.  This was just
>  an oversight when this feature was added.
>  
>  * Add CALL-IDs when using syslog
>  
>  (closes issue ASTERISK-21430)
>  Reported by: Nikola Ciprich
>  Tested by: Nikola Ciprich, Michael L. Young
>  Patches:
>      asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026)
>  
>  Review: https://reviewboard.asterisk.org/r/2526/
>  ........
>  
>  Merged revisions 388605 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388668 | kmoore | 2013-05-14 09:47:52 -0300 (Tue, 14 May 2013) | 10 lines
>  
>  Move JSON event generators into separate modules
>  
>  This moves the JSON event generators out of the Stasis-HTTP modules and
>  into standalone JSON-related counterparts so that Stasis-HTTP and
>  res_stasis can depend on them without creating dependency cycles. This
>  also provides a future location for Swagger Model validator functions
>  once the generators for that code are written.
>  
>  Review: https://reviewboard.asterisk.org/r/2534/
>........
>  r388701 | rmudgett | 2013-05-14 16:03:26 -0300 (Tue, 14 May 2013) | 14 lines
>  
>  Make ao2 global objects not always use the debug version of the ao2_ref() calls.
>  
>  The debug versions of ao2_ref() should only be used if REF_DEBUG is
>  enabled so nothing is written to /tmp/refs unexpectedly.
>  
>  (closes issue ASTERISK-21785)
>  Reported by: abelbeck
>  Patches:
>        jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett
>  Tested by: abelbeck
>  ........
>  
>  Merged revisions 388700 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388729 | dlee | 2013-05-14 18:45:08 -0300 (Tue, 14 May 2013) | 20 lines
>  
>  Break res_stasis into smaller files.
>  
>  When implementing playback for stasis-http, the monolithicedness of
>  res_stasis really started to get in my way.
>  
>  This patch breaks the major components of res_stasis.c into individual
>  files.
>  
>   * res/stasis/app.c - Stasis application tracking
>   * res/stasis/control.c - Channel control objects
>   * res/stasis/command.c - Channel command object
>  
>  This refactoring also allows res_stasis applications to be loaded as
>  independent modules, such as the new res_stasis_answer module.
>  
>  The bulk of this patch is simply moving code from one file to another,
>  adjusting names and adding accessors as necessary.
>  
>  Review: https://reviewboard.asterisk.org/r/2530/
>........
>  r388751 | dlee | 2013-05-14 23:37:22 -0300 (Tue, 14 May 2013) | 3 lines
>  
>  Refactored the rest of the message types to use the STASIS_MESSAGE_TYPE_*
>  macros.
>........
>  r388770 | kmoore | 2013-05-15 09:42:04 -0300 (Wed, 15 May 2013) | 14 lines
>  
>  Use srtp_shutdown when available
>  
>  This allows the SRTP library to be shut down properly when the
>  functionality is offered by libsrtp.
>  
>  Review: https://reviewboard.asterisk.org/r/2538/
>  (closes issue ASTERISK-21719)
>  ........
>  
>  Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 388769 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388818 | qwell | 2013-05-15 12:03:40 -0300 (Wed, 15 May 2013) | 18 lines
>  
>  Fix VM snapshot handling for combined INBOX.
>  
>  The snapshot API contains an option that allow for combining of new 
>  and old messages within a single snapshot. New messages, however, 
>  include options beyond just 'INBOX' - it also includes the Urgent 
>  folder. A previous patch that combined INBOX and Urgent accidentally 
>  impacted snapshots that attempted to gain messages from just the Old 
>  folder. This patch fixes the snapshot gathering such that the API 
>  returns the appropriate messages for the folder selected, with and 
>  without the combine option.
>  
>  This should make it more clear about what's happening.
>  
>  Review: https://reviewboard.asterisk.org/r/2539/
>  ........
>  
>  Merged revisions 388816 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388840 | kharwell | 2013-05-15 12:58:56 -0300 (Wed, 15 May 2013) | 18 lines
>  
>  Fix for segfault in __ast_rwlock_destroy with DEBUG_THREADS
>  
>  If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying
>  to access a possible NULL t->track object.  A NULL check has been added before
>  trying to access the memory.
>  
>  (closes issue ASTERISK-21724)
>  Reported by: Corey Farrell
>  Fixed by: Corey Farrell
>  Patches:
>  	ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909)
>  ........
>  
>  Merged revisions 388838 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>  ........
>  
>  Merged revisions 388839 from http://svn.asterisk.org/svn/asterisk/branches/11
>........
>  r388896 | dlee | 2013-05-15 18:13:29 -0300 (Wed, 15 May 2013) | 4 lines
>  
>  Fixed inverted logic in app_add_channel().
>  
>  Also added some missing doc comments for stasis/app.h.
>........
>  r388975 | jrose | 2013-05-17 14:36:10 -0300 (Fri, 17 May 2013) | 10 lines
>  
>  Stasis: Update security events to use Stasis
>  
>  Also moves ACL messages to the security topic and gets rid of the
>  ACL topic
>  
>  (closes issue ASTERISK-21103)
>  Reported by: Matt Jordan
>  Review: https://reviewboard.asterisk.org/r/2496/
>........
>  r388976 | mjordan | 2013-05-17 14:43:58 -0300 (Fri, 17 May 2013) | 19 lines
>  
>  Publish the outbound channel's application/data when dialing
>  
>  This patch does two things:
>  * It fixes a bug where the outbound channel's application/data set by the
>    dialing API/app_dial is not communicated until the channel is hung up.
>    If that happens, AMI would incorrectly send a NewExten event immediately
>    after a Hangup. This isn't really AMI's fault, as the dialing APIs never
>    communicated the 'helpful' app/data on the outbound channel until it was
>    hungup.
>  * It makes public sending a stasis message about a change in channel state.
>    This is useful enough that - for now at least - it should be public. If
>    operations on a channel go to being more coarse-grained, this function
>    could be made private again.
>  
>  Review: https://reviewboard.asterisk.org/r/2548
>  
>  Note that this problem was found and reported by Matt DiMeo.
>........
>  r389009 | elguero | 2013-05-17 17:24:56 -0300 (Fri, 17 May 2013) | 17 lines
>  
>  Remove Character Limit On "inkeys" For IAX2
>  
>  Currently, the buffer for processing "inkeys" is limited to 256 characters.  If
>  the user has many keys and the names of those key files are long, the 256
>  character limit is not enough.
>  
>  * Change inkeys buffer to be dynamic
>  
>  (closes issue ASTERISK-21398)
>  Reported by: Pavel Kopchyk
>  Tested by: Pavel Kopchyk, Michael L. Young
>  Patches:
>      asterisk-21398-iax2-inkeys-dynamic-buffer_v3.diff
>  					by Michael L. Young (license 5026)
>  
>  Review: https://reviewboard.asterisk.org/r/2501/
>........
>  r389011 | dlee | 2013-05-17 18:10:32 -0300 (Fri, 17 May 2013) | 27 lines
>  
>  Fix shutdown assertions in stasis-core
>  
>  In r388005, macros were introduced to consistently define message
>  types. This added an assert if a message type was used either before
>  it was initialized or after it had been cleaned up. It turns out that
>  this assertion fires during shutdown.
>  
>  This actually exposed a hidden shutdown ordering problem. Since
>  unsubscribing is asynchronous, it's possible that the message types
>  used by the subscription could be freed before the final message of
>  the subscription was processed.
>  
>  This patch adds stasis_subscription_join(), which blocks until the
>  last message has been processed by the subscription. Since joining was
>  most commonly done right after an unsubscribe, a
>  stasis_unsubscribe_and_join() convenience function was also added.
>  
>  Similar functions were also added to the stasis_caching_topic and
>  stasis_message_router, since they wrap subscriptions and have similar
>  problems.
>  
>  Other code in trunk was refactored to join() where appropriate, or at
>  least verify that the subscription was complete before being
>  destroyed.
>  
>  Review: https://reviewboard.asterisk.org/r/2540
>........
>  r389053 | file | 2013-05-18 16:47:24 -0300 (Sat, 18 May 2013) | 7 lines
>  
>  Move origination to use the dialing API and send Stasis messages on dial begin and end.
>  
>  (closes issue ASTERISK-21549)
>  Reported by: Matt Jordan
>  
>  Review: https://reviewboard.asterisk.org/r/2512/
>........
>  r389085 | file | 2013-05-18 19:49:14 -0300 (Sat, 18 May 2013) | 2 lines
>  
>  Fix a bug where synchronous origination (oddly enough triggered by doing an async manager Originate) would not work properly.
>........
>  r389097 | wedhorn | 2013-05-18 20:20:53 -0300 (Sat, 18 May 2013) | 16 lines
>  
>  Add call forward no answer to skinny and cleanup general callfwd handling.
>  
>  CallforwardNoAnswer uses a sched to determine when to forward the call. 
>  Defaults to 20secs but configurable in skinny.conf.
>  
>  Adds dialType to each subchannel structure to be used to differentiate
>  between normal dials that result in a call being placed (default) and
>  other uses for the skinny_dialer (such as cfwd digit collection).
>  Restructured all cfwd handling to use this new arrangement.
>  
>  (closes issue ASTERISK-21292)
>  Reported by: wedhorn
>  Tested by: myself
>  Patches: 
>      skinny-callfwdnoans03.diff uploaded by wedhorn (license 5019)
>........
>  r389116 | file | 2013-05-18 21:49:15 -0300 (Sat, 18 May 2013) | 2 lines
>  
>  If the caller of the originate API calls wants the channel ensure it has been requested and dialed.
>........
>  r389132 | file | 2013-05-18 23:21:44 -0300 (Sat, 18 May 2013) | 2 lines
>  
>  Don't hold the outgoing lock for a prolonged period of time as it may block the originator.
>........
>  r389148 | kmoore | 2013-05-19 14:45:42 -0300 (Sun, 19 May 2013) | 9 lines
>  
>  Add base XML documentation for res_sip
>  
>  Thanks to Brad Latus, this patch adds a significant amount much-needed
>  documentation to res_sip. It should cover all existing configuration
>  options currently in Asterisk trunk.
>  
>  Patch-by: Brad Latus (snuffy)
>  Review: https://reviewboard.asterisk.org/r/2471/
>........
>  r389164 | wedhorn | 2013-05-19 16:45:14 -0300 (Sun, 19 May 2013) | 8 lines
>  
>  Add transfer softkey to ringout state to enable blond transfers.
>  
>  (closes issue ASTERISK-21327)
>  Reported by: wedhorn
>  Tested by: myself
>  Patches: 
>      skinny-blindxfer01.diff uploaded by wedhorn (license 5019)
>........
>  r389180 | may | 2013-05-19 17:52:34 -0300 (Sun, 19 May 2013) | 2 lines
>  
>  add ast_publish_channel_state according new event framework
>........
>
>Merged revisions 388526,388531,388579,388598,388602,388617,388668,388701,388729,388751,388770,388818,388840,388896,388975-388976,389009,389011,389053,389085,389097,389116,389132,389148,389164,389180 from http://svn.asterisk.org/svn/asterisk/trunk
>


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