[svn-commits] bebuild: tag 1.8.31.0-rc1 r423583 - /tags/1.8.31.0-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Fri Sep 19 15:15:00 CDT 2014


Author: bebuild
Date: Fri Sep 19 15:14:57 2014
New Revision: 423583

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423583
Log:
Importing files for 1.8.31.0-rc1 release.

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--- tags/1.8.31.0-rc1/ChangeLog (added)
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+2014-09-19  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.31.0-rc1 Released.
+
+2014-09-18 16:08 +0000 [r423349]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/astobj2.c, contrib/scripts/refcounter.py:
+	  astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
+	  Make astob2 REF_DEBUG output an invalid object line when an
+	  invalid ao2 object ref/unref is attempted. This is similar to the
+	  constructor/destructor lines. * Fixed refcounter.py to handle
+	  skewed objects that have constructor/destructor states. * Made
+	  refcounter.py highlight the invalid ao2 object refs by putting
+	  them in their own section of the processed output file. * Made
+	  refcounter.py highlight unreffing an object by more than one that
+	  results in a negative ref count and the object being destroyed.
+	  The abnormally destroyed object is reported in the invalid and
+	  finalized object sections of the output. Review:
+	  https://reviewboard.asterisk.org/r/3971/
+
+2014-09-18 14:37 +0000 [r423276]  George Joseph <george.joseph at fairview5.com>
+
+	* main/config.c, main/manager.c, include/asterisk/config.h: config:
+	  bug: Fix SEGV in ast_category_insert when matching category isn't
+	  found If you call ast_category_insert with a match category that
+	  doesn't exist, the list traverse runs out of 'next' categories
+	  and you get a SEGV. This patch adds check for the end-of-list
+	  condition and changes the signature to return an int for
+	  success/failure indication instead of a void. The only consumer
+	  of this function is manager and it was also changed to use the
+	  return value. Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3993/
+
+2014-09-14 15:48 +0000 [r423066]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* configs/sip.conf.sample: chan_sip: Clarify that sipdebug=yes
+	  cannot be undone by the CLI. Document it in sip.conf.
+	  ASTERISK-24249 #close Reported by: Avinash Mohod Review:
+	  https://reviewboard.asterisk.org/r/3926/
+
+2014-09-12 18:17 +0000 [r423006]  Kinsey Moore <kmoore at digium.com>
+
+	* main/channel.c: Bridging: Fix bouncing native bridge This fixes a
+	  situation in Asterisk 1.8 and 11 where ast_channel_bridge could
+	  cause a bouncing native bridge. In the case of the
+	  dial_LS_options test, this was a remote RTP bridge which caused
+	  the audio path to continually cycle between Asterisk and the
+	  remote endpoints generating a large number of SIP messages and
+	  delaying the test long enough to cause it to fail (checking
+	  timing was part of the test). The root cause was that the code to
+	  decide whether to use native bridging was expecting a
+	  time-remaining value of 0 to be the default instead of the actual
+	  default value of -1. A value of 0 or negative numbers could also
+	  be generated by preceding code in some circumstances. Both issues
+	  are addressed in this patch. ASTERISK-24211 #close Reported by:
+	  Matt Jordan Review: https://reviewboard.asterisk.org/r/3987/
+
+2014-09-10 15:58 +0000 [r422900]  George Joseph <george.joseph at fairview5.com>
+
+	* main/config.c: config: bug: fix truncation of included config
+	  files on permissions error ast_config_text_file_save() currently
+	  truncates include files as they are processed. If a subsequent
+	  include file or the main config file has a permissions error that
+	  prevents writing, earlier include files are left truncated
+	  resulting in a frantic search for backups. This patch causes
+	  ast_config_text_file_save to check for write access on all files
+	  before it truncates any of them. Will be applied 1.8 > trunk.
+	  Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3986/
+
+2014-09-07 00:07 +0000 [r422789]  Rusty Newton <rnewton at digium.com>
+
+	* sounds/Makefile, sounds/sounds.xml: Sounds/BuildSystem:
+	  Modifications to include new releases and Japanese language.
+	  Modifying Makefile and sounds.xml to include new core 1.4.26 and
+	  extra 1.4.15 sound prompt releases, plus the new Japanese core
+	  sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
+	  Kevin McCoy Tested by: Rusty Newton
+
+2014-09-04 19:51 +0000 [r422584]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c: Manager: Require read permission for SYSTEM in
+	  order to send FullyBooted Review:
+	  https://reviewboard.asterisk.org/r/3969/
+
+2014-08-30 17:19 +0000 [r422439]  George Joseph <george.joseph at fairview5.com>
+
+	* main/manager.c: manager: Make WaitEvent action respect
+	  eventfilters A WaitEvent issued via an http session isn't
+	  respecting eventfilters defined for the user. I just added a
+	  match_filter to the predicate that controls astman_append. Tested
+	  by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3958/
+
+2014-08-29 19:38 +0000 [r422293-422376]  Matthew Jordan <mjordan at digium.com>
+
+	* doc/smsq.8 (added): doc: Add a manpage for the smsq utility This
+	  patch adds a manpage for the smsq utility. Note that this is one
+	  of the patches the Debian distro applies for the Asterisk
+	  project, as per ASTERISK-24191. Review:
+	  https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close
+	  Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy
+	  Laine (License 6561)
+
+	* doc/aelparse.8 (added): doc: Add a manpage for the aelparse
+	  utility This patch adds a manpage for the aelparse utility. Note
+	  that this is one of the patches the Debian distro applies for the
+	  Asterisk project, as per ASTERISK-24191. Review:
+	  https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close
+	  Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy
+	  Laine (License 6561)
+
+	* LICENSE: LICENSE: Clarify language in Asterisk's LICENSE to allow
+	  for linking to UniMRCP The UniMRCP project distributes Asterisk
+	  modules that integrate Asterisk with UniMRCP, and other Asterisk
+	  users use the UniMRCP library as well. Unfortunately, the UniMRCP
+	  license is Apache 2.0, which per the Free Software Foundation, is
+	  not a compatible license with the GPLv2. "Please note that this
+	  license is not compatible with GPL version 2, because it has some
+	  requirements that are not in that GPL version. These include
+	  certain patent termination and indemnification provisions. The
+	  patent termination provision is a good thing, which is why we
+	  recommend the Apache 2.0 license for substantial programs over
+	  other lax permissive licenses." On the other hand, UniMRCP is a
+	  great project and we'd like to let people use it with Asterisk.
+	  This patch updates the LICENSE text to allow users to link
+	  Asterisk with UniMRCP and distribute the resulting binaries.
+
+2014-08-27 14:25 +0000 [r422112]  Kinsey Moore <kmoore at digium.com>
+
+	* include/asterisk/utils.h, channels/chan_sip.c,
+	  tests/test_callerid.c (added), tests/test_utils.c,
+	  main/callerid.c, main/utils.c: CallerID: Fix parsing of malformed
+	  callerid This allows the callerid parsing function to handle
+	  malformed input strings and strings containing escaped and
+	  unescaped double quotes. This also adds a unittest to cover many
+	  of the cases where the parsing algorithm previously failed.
+	  Review: https://reviewboard.asterisk.org/r/3923/
+
+2014-08-25 16:00 +0000 [r421976]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_musiconhold.c: res_musiconhold: Fix MOH restarting where
+	  it left off from the last hold. Restore code removed by
+	  https://reviewboard.asterisk.org/r/3536/ that introduced a
+	  regression that prevents MOH from restarting were it left off the
+	  last time. ASTERISK-24019 #close Reported by: Jason Richards
+	  Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch
+	  uploaded by rmudgett Review:
+	  https://reviewboard.asterisk.org/r/3928/
+
+2014-08-21 22:01 +0000 [r421799]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_musiconhold.c: res_musiconhold.c: Remove obsolete
+	  REF_DEBUG code. Remove unneeded code that writes to the wrong
+	  file location in an obsolete format.
+
+2014-08-21 17:32 +0000 [r421717]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Don't use port derived from
+	  fromdomain if it isn't set If a user does not provide a port in
+	  the fromdomain setting, chan_sip will set the fromdomainport to
+	  STANDARD_SIP_PORT (5060). The fromdomainport value will then get
+	  used unilaterally in certain places. This causes issues with TLS,
+	  where the default port is expected to be 5061. This patch
+	  modifies chan_sip such that fromdomainport is only used if it is
+	  not the standard SIP port; otherwise, the port from the SIP pvt's
+	  recorded self IP address is used. Review:
+	  https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close
+	  Reported by: Elazar Broad patches: fromdomainport_fix.diff
+	  uploaded by Elazar Broad (License 5835)
+
+2014-08-20 22:13 +0000 [r421600]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/cli.c: cli.c: Fix tab completion of "module load" when
+	  MALLOC_DEBUG is enabled. filename_completion_function() returns
+	  memory that was not allocated by the MALLOC_DEBUG allocation
+	  tracker so the memory must be freed by ast_std_free().
+
+2014-08-19 19:38 +0000 [r421442]  Kinsey Moore <kmoore at digium.com>
+
+	* main/manager.c: AMI Docs: Fix Status channel parameter
+	  optionality
+
+2014-08-18 20:14 +0000 [r421327]  George Joseph <george.joseph at fairview5.com>
+
+	* funcs/func_config.c: func_config: Change 'Not Found' message from
+	  ERROR to DEBUG When you call the CONFIG dialplan function with
+	  the name of a variable that doesn't exist in the target context
+	  you get an ERROR. This does nothing but clutter up the logs with
+	  messages that may be perfectly acceptable. Just because a
+	  variable wasn't in the context doesn't mean it's an error. Maybei
+	  t's optional or just needs to be defaulted or ignored. This patch
+	  changes the log level from ERROR to DEBUG. If a dialplan
+	  developer wants to debug their dialplan they still canby setting
+	  the console debug level as needed. Tested by: George Joseph
+	  Review: https://reviewboard.asterisk.org/r/3919/
+
+2014-08-17 23:06 +0000 [r421227-421232]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_dial.c: apps/app_dial: Fix Dial 'z' option The 'z'
+	  option is supposed to disable the dial timeout in the case of a
+	  call forward. Unfortunately, the wrong timeout timer was passed
+	  to the do_forward function, resulting in the option not working.
+	  ASTERISK-24225 #close Reported by: dimitripietro Tested by:
+	  dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by
+	  rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by
+	  rmudgett (License 5621)
+
+	* configure, configure.ac: configure: Undefine FORTIFY_SOURCE prior
+	  to defining it for patched gcc Some distributions of Linux patch
+	  gcc to define FORTIFY_SOURCE when gcc is executed with
+	  optimization. This "help" unfortunately results in re-definition
+	  warnings when FORTIFY_SOURCE is later defined in Asterisk's build
+	  system. This patch undefines FORTIFY_SOURCE prior to defining it
+	  to prevent this warning. Review:
+	  https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close
+	  Reported by: Kilburn Tested by: Kilburn, wdoekes patches:
+	  1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by
+	  cloos (License 5956) 11.diff uploaded by cloos (License 5956)
+	  12.diff uploaded by cloos (License 5956) 13.diff uploaded by
+	  cloos (License 5956)
+
+2014-08-15 14:43 +0000 [r421059-421125]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_voicemail.c, main/app.c: app_voicemail/app: Remove test
+	  events that were duplicated by r421059 Moving the test event
+	  raised when a file is played back (which occurred in r421059)
+	  broke the ever loving snot out of the voicemail tests. This
+	  caused duplicate test events to get raised, as app_voicemail and
+	  main/app were raising events prior to call ast_streamfile. The
+	  voicemail tests did not enjoy getting multiple events. Since
+	  raising the playback event in ast_streamfile is far more useful
+	  to the vast majority of tests, this patch keeps the call there
+	  and simply removes the extraneous calls that duplicated the
+	  event.
+
+	* main/file.c: main/file: Move test event to emit PLAYBACK event
+	  more consistently This is being done in advance of the test for
+	  ASTERISK-23953
+
+2014-08-19  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.30.0 Released.
+
+2014-08-11  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.30.0-rc1 Released.
+
+2014-08-11 10:24 +0000 [r420654-420680]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* main/utils.c: general: Fix memory Corruption in
+	  __ast_string_field_ptr_build_va. If the space left in a
+	  stringfield is between 0 and
+	  (alignof(ast_string_field_allocation)-1) adding new data would
+	  cause memory corruption, because we would assume enough space
+	  (unsigned underrun). Thanks Arnd Schmitter for reporting and
+	  finding out the cause! ASTERISK-23508 #close Reported by: Arnd
+	  Schmitter Tested by: Arnd Schmitter, JoshE Review:
+	  https://reviewboard.asterisk.org/r/3898/
+
+	* main/tcptls.c: tcptls: Avoid compiler warning on non-dev-mode.
+
+2014-08-07 21:25 +0000 [r420434]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Replace sip_tls_read() and resolve
+	  the large SDP poll issue. Replace sip_tls_read() and
+	  sip_tcp_read() with a single function and resolve the poll/wait
+	  issue with large SDP payloads. ASTERISK-18345 #close Reported by:
+	  Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
+	  patch uploaded by Elazar Broad Review:
+	  https://reviewboard.asterisk.org/r/3882/
+
+2014-08-06 16:05 +0000 [r420146]  George Joseph <george.joseph at fairview5.com>
+
+	* pbx/pbx_lua.c, main/pbx.c: pbx_lua: fix regression with global
+	  sym export and context clash by pbx_config. ASTERISK-23818 (lua
+	  contexts being overwritten by contexts of the same name in
+	  pbx_config) surfaced because pbx_lua, having the
+	  AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before
+	  pbx_config. Since I couldn't find any reason for pbx_lua to
+	  export it's symbols to the rest of Asterisk, I simply changed the
+	  flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
+	  realize was that the symbols need to be exported not because
+	  Asterisk needs them but because any external Lua modules like
+	  luasql.mysql need the base Lua language APIs exported
+	  (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's
+	  an issue in pbx.c where context_merge was only merging includes,
+	  switches and ignore patterns if the context was already existing
+	  AND has extensions, or if the context was brand new. If pbx_lua
+	  is loaded before pbx_config, the context will exist BUT pbx_lua,
+	  being implemented as a switch, will never place extensions in it,
+	  just the switch statement. The result is that when pbx_config
+	  loads, it never merges the switch statement created by pbx_lua
+	  into the final context. This patch sets pbx_lua's modflag back to
+	  AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge
+	  that catches the case where an existing context has includes,
+	  switchs or ingore patterns but no actual extensions.
+	  ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo
+	  Teräs Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3891/
+
+2014-08-04 19:42 +0000 [r419942]  Rusty Newton <rnewton at digium.com>
+
+	* main/manager.c: Manager - Improve documentation for manager
+	  commands Getvar and Setvar. The documentation for these commands
+	  did not make it clear that they could accept expressions and
+	  functions. Modified to make this clear, but tried not to be
+	  overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton
+	  Tested by: Rusty Newton Review:
+	  https://reviewboard.asterisk.org/r/3854
+
+2014-07-28 18:27 +0000 [r419684]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_queue.c, apps/app_speech_utils.c,
+	  funcs/func_frame_trace.c: datastores: Audit
+	  ast_channel_datastore_remove usage. Audit of v1.8 usage of
+	  ast_channel_datastore_remove() for datastore memory leaks. *
+	  Fixed leaks in app_speech_utils and func_frame_trace. * Fixed
+	  app_speech_utils not locking the channel when accessing the
+	  channel datastore list. Review:
+	  https://reviewboard.asterisk.org/r/3859/
+
+2014-07-25 23:04 +0000 [r419630]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/app.h, main/features.c, main/app.c,
+	  apps/app_stack.c: features.c: Allow appliationmap to use Gosub.
+	  Using DYNAMIC_FEATURES with a Gosub application as the mapped
+	  application does not work. It does not work because Gosub just
+	  pushes the current dialplan context, exten, and priority onto a
+	  stack and sets the specified Gosub location. Gosub does not have
+	  a dialplan execution loop to run dialplan like Macro. * Made the
+	  DYNAMIC_FEATURES application mapping feature call
+	  ast_app_exec_macro() and ast_app_exec_sub() for the Macro and
+	  Gosub applications respectively. * Backported
+	  ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute
+	  dialplan routines from the DYNAMIC_FEATURES application mapping
+	  feature. NOTE: This issue does not affect v12+ because it already
+	  does what this patch implements. AST-1391 #close Reported by:
+	  Guenther Kelleter Review:
+	  https://reviewboard.asterisk.org/r/3844/
+
+2014-07-24 17:55 +0000 [r419440]  Corey Farrell <git at cfware.com>
+
+	* channels/chan_sip.c: chan_sip: sip_subscribe_mwi_destroy should
+	  not call sip_destroy sip_subscribe_mwi_destroy calls sip_destroy
+	  on the reference counted mwi->call. This results in the fields of
+	  mwi->call being freed, but mwi->call itself it leaked. If other
+	  code is still using mwi->call it can cause problems. This change
+	  uses dialog_unref instead, to balance the ref provided by
+	  sip_alloc(). ASTERISK-24087 #close Reported by: Corey Farrell
+	  Review: https://reviewboard.asterisk.org/r/3834/
+
+2014-07-24 16:47 +0000 [r419374]  Jason Parker <jparker at digium.com>
+
+	* addons/chan_ooh323.c: Don't cause Asterisk to exit if ooh323.conf
+	  not found. (closes issue ASTERISK-23814)
+
+2014-07-22 13:17 +0000 [r419129]  Kinsey Moore <kmoore at digium.com>
+
+	* addons/chan_ooh323.c, tests/test_astobj2_thrash.c,
+	  apps/app_meetme.c, tests/test_logger.c,
+	  addons/ooh323c/src/decode.c, tests/test_event.c,
+	  tests/test_aoc.c, tests/test_hashtab_thrash.c,
+	  tests/test_astobj2.c, addons/ooh323c/src/printHandler.c,
+	  addons/ooh323c/src/ooq931.c: Fix more dev-mode build issues
+
+2014-07-15 17:19 +0000 [r418641]  Jonathan Rose <jrose at digium.com>
+
+	* funcs/func_uri.c: func_uri: URIENCODE/URIDECODE - allow empty
+	  strings as argument Previously these two dialplan functions would
+	  issue warnings and return failure when an empty string is used as
+	  the argument. Now they will not issue a warning and will
+	  successfully return an empty string. ASTERISK-23911 #close
+	  Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3745/
+
+2014-07-13 21:47 +0000 [r418504]  Corey Farrell <git at cfware.com>
+
+	* main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work
+	  around REF_DEBUG race which causes out of order log entries *
+	  Update refcounter.py to use delta's to track the current
+	  reference count. * Use result from internal_ao2_ref to write
+	  old_refcount to refs_log. Review:
+	  https://reviewboard.asterisk.org/r/3756/
+
+2014-07-10 01:23 +0000 [r418261]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c: chan_dahdi/sig_pri: Fix type mismatch in the
+	  idledial feature's channel creation. Square pegs in round holes
+	  don't work very well.
+
+2014-07-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.29.0 Released.
+
+2014-07-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.29.0-rc1 Released.
+
+2014-07-03 21:38 +0000 [r417956]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+	  configs/chan_dahdi.conf.sample, UPGRADE.txt: chan_dahdi: Add
+	  inband_on_setup_ack compatibility option. The new
+	  inband_on_setup_ack option causes Asterisk to assume inband audio
+	  may be present when a SETUP_ACKNOWLEDGE message is received.
+	  Q.931 Section 5.1.3 says that in scenarios with overlap dialing,
+	  when a dialtone is sent from the network side, progress indicator
+	  8 "Inband info now available" MAY be sent to the CPE if no digits
+	  were received with the SETUP. It is thus implied that the ie is
+	  mandatory if digits came with the SETUP and dialtone is needed.
+	  This option should be enabled, when the network sends dialtone
+	  and you want to hear it, but the network doesn't send the
+	  progress indicator when needed. NOTE: For Q.SIG setups this
+	  option should be enabled when outgoing overlap dialing is also
+	  enabled because Q.SIG does not send the progress indicator with
+	  the SETUP ACK. The commit -r413714 (AST-1338) which causes this
+	  issue was dealing with a SIP-to-ISDN interoperability issue. This
+	  commit is a merge of the two patches indicated below.
+	  ASTERISK-23897 #close Reported by: Pavel Troller Patches:
+	  pri-4.diff (license #6302) patch uploaded by Pavel Troller
+	  jira_asterisk_23897_v11.patch (license #5621) patch uploaded by
+	  rmudgett Review: https://reviewboard.asterisk.org/r/3633/
+
+2014-07-03 11:19 +0000 [r417797]  Matthew Jordan <mjordan at digium.com>
+
+	* main/utils.c: main/untils: Prevent potential infinite loop in
+	  ast_careful_fwrite A loop in ast_careful_fwrite exists that will
+	  continually attempt to write to a file stream, even in the
+	  presence of EAGAIN/EINTR errors. However, if a connection that
+	  uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
+	  call to fflush may return EAGAIN/EINTER along with EOF. A
+	  subsequent call to fflush will return EOF but not clear errno,
+	  resulting in an infinite loop. This patch clears errno after it
+	  is detected and handled the loop, such that any subsequent call
+	  to fflush will not get erroneously stuck. Review:
+	  https://reviewboard.asterisk.org/r/3704 ASTERISK-23984 #close
+	  Reported by: Steve Davies patches: fflush_loop_fix uploaded by
+	  one47 (License 5012)
+
+2014-06-30 03:20 +0000 [r417587]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: chan_sip: be more tolerant of whitespace
+	  between attributes in SDP fmtp line This patch is essentially a
+	  backport of a small portion of r397526 from ASTERISK-21981. In
+	  that patch, pass through support and format attribute negotiation
+	  was added for Opus. Part of that included being more tolerant to
+	  whitespace in the fmtp line of an SDP; that part of the patch is
+	  being applied here. As the author of the backport pointed out, in
+	  SDP, the fmtp line is allowed to include whitespace between
+	  attributes. RFC 3267 chapter 8.3 (from 2001) includes an example
+	  for this. This was not removed in the updated RFC 4867 in 2007.
+	  Note that this patch only applies to audio in Asterisk 1.8, which
+	  is a bit more limited in its support for format attributes. It
+	  does have limited support for some codecs, so this patch is still
+	  useful in this version. Review:
+	  https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 Reported
+	  by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch
+	  uploaded by Alexander Traud (License 6520)
+
+2014-06-27 19:24 +0000 [r417480-417500]  Corey Farrell <git at cfware.com>
+
+	* main/astobj2.c: Ensure REF_DEBUG records entrys for attempts to
+	  ao2_ref an invalid object This change ensures that
+	  __ao2_ref_debug writes to ref_log when given a non-NULL pointer
+	  to an invalid ao2 object. This is to ensure that we record any
+	  attempt manipulate references of already freed objects.
+	  ASTERISK-23948 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3677/
+
+	* contrib/scripts/refcounter.py: refcounter.py: prevent use of
+	  excessive RAM with large refs logs When processing a 212MB refs
+	  file, refcounter.py used over 3GB of RAM. This change greatly
+	  reduces memory usage in two ways: * Saving object history in
+	  whole lines instead of separated values. * Not saving
+	  normal/skewed/leaked object lists unless they are requested.
+	  ASTERISK-23921 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3668/
+
+2014-06-26 12:21 +0000 [r417318]  Matthew Jordan <mjordan at digium.com>
+
+	* main/udptl.c: udptl: Correct FEC to not consider negative
+	  sequence numbers as missing When using FEC, with span=3 and
+	  entries=4 Asterisk will attempt to repair the packet with
+	  sequence number 5, as it will see that packet -4 is missing. The
+	  result is Asterisk sending garbage packets that can kill a fax.
+	  This patch adds a check to see if the sequence number is valid
+	  before checking if the packet is missing. Review:
+	  https://reviewboard.asterisk.org/r/3657/ ASTERISK-23908 #close
+	  Reported by: Torrey Searle patches: udptl_fec.patch uploaded by
+	  Torrey Searle (License 5334)
+
+2014-06-26 10:02 +0000 [r417248]  Corey Farrell <git at cfware.com>
+
+	* channels/chan_sip.c: chan_sip: Fix handling of "From" headers
+	  longer than 256 characters From headers were processed using a
+	  256 character buffer on the stack. This change replaces that with
+	  a heap allocation by ast_strdup. ASTERISK-23790 #close Reported
+	  by: uniken1 Tested by: uniken1 Review:
+	  https://reviewboard.asterisk.org/r/3669/ Patches:
+	  chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes
+	  (license 5674)
+
+2014-06-23 14:34 +0000 [r417076]  Rusty Newton <rnewton at digium.com>
+
+	* configs/features.conf.sample: main/features - documentation -
+	  reformat examples and options in features.conf.sample to show
+	  clearly which options apply in which section The features.conf
+	  sample can be a bit confusing about what parking options can be
+	  set only in the general context, or both in the general context
+	  (for the default parking lot) and in other parking lot contexts.
+	  A bug was filed due to confusion and a little googling will show
+	  lots of other confused users. Despite some comments on the
+	  individual options, it still reads in a confusing way. In this
+	  patch I separate out those options with some headings in to
+	  attempt a better layout. I went ahead and modified other headings
+	  in the file, or added them to facilitate better visual scanning.
+	  ASTERISK-23667 #close Review:
+	  https://reviewboard.asterisk.org/r/3621/
+
+2014-06-22 20:46 +0000 [r417016]  George Joseph <george.joseph at fairview5.com>
+
+	* Makefile.rules, Makefile: build: Turn FORTIFY_SOURCE off if
+	  DONT_OPTIMIZE is set. AST_FORTIFY_SOURCE is automatically set in
+	  ./Makefile even if DONT_OPTIMIZE is set in menuselect. This
+	  causes gcc to complain that _FORTIFY_SOURCE requires optimization
+	  and the build will fail. You can specify "make
+	  AST_FORTIFY_SOURCE=''" but I always forget. This patch moves the
+	  set of AST_FORTIFY_SOURCE to Makefile.rules and only sets it if
+	  DONT_OPTIMIZE is "no". The move is necessary because the
+	  top-level Makefile doesn't include menuselect.makeopts. This
+	  doesn't solve the entire problem however because res_config_mysql
+	  seems to force _FORTIFY_SOURCE so res_config_mysql has to be
+	  disabled for now if DONT_OPTIMIZE is set. Tested by: George
+	  Joseph Review: https://reviewboard.asterisk.org/r/3664/
+
+2014-06-20 23:12 +0000 [r416869-416929]  George Joseph <george.joseph at fairview5.com>
+
+	* configure, include/asterisk/autoconfig.h.in: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. A regenerated ./configure and
+	  include/asterisk/autoconfig.h.in are included but can be
+	  regenerated by running ./bootstrap.sh at any time. Tested by:
+	  George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/
+
+	* autoconf/ast_ext_tool_check.m4: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. Tested by: George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/
+
+2014-06-19 19:33 +0000 [r416732]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/sip/reqresp_parser.c, main/test.c: Fix build warnings
+	  with TEST_FRAMEWORK enabled
+
+2014-06-19 15:59 +0000 [r416578-416667]  George Joseph <george.joseph at fairview5.com>
+
+	* pbx/pbx_lua.c: Remove the problematic and unneeded
+	  AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
+	  AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be
+	  incorrectly loaded before pbx_config. pbx_config was therefore
+	  blowing away contexts that were created by pbx_lua. With
+	  AST_MODFLAG_DEFAULT the load order is now correct and contexs are
+	  being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed
+	  anyway since no other modules needed its global symbols that
+	  early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by:
+	  Dennis Guse Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3629/
+
+	* configs/extensions.lua.sample: Update extensions.lua.sample with
+	  naming conflict guidance. The sample extensions.lua was causing
+	  pbx_lua to fail to load when parsing 'app.goto("default", "s",
+	  1)' because in Lua 5.2, 'goto' is now a reserved word. This patch
+	  adds guidance to extensions.lua.sample and changed
+	  'app.goto("default", "s", 1)' to 'app.['goto']("default", "s",
+	  1)'. https://reviewboard.asterisk.org/r/3627/ ASTERISK-23844
+	  #comment This commit fixes 1.8, patch for 11->trunk coming.
+
+2014-06-17 18:22 +0000 [r416500]  Mark Michelson <mmichelson at digium.com>
+
+	* funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to set
+	  inheritable channel variables.
+
+2014-06-17 16:20 +0000 [r416439]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_musiconhold.c: MoH: Don't restart stream on repeated
+	  start calls Currently, music on hold will stop and then start
+	  again from the beginning if ast_moh_start() is called multiple
+	  times. This can happen if a call is put on hold repeatedly (the
+	  channel receives multiple HOLD control frames) and can be
+	  triggered from ARI by starting MoH on a channel multiple times.
+	  This is fairly jarring/annoying to users. This change prevents
+	  MoH from being restarted if the requested music class is the same
+	  as the one currently playing. This includes an extra check to
+	  prevent the errors previously experienced in the testsuite and
+	  has 100+ test runs behind it. Review:
+	  https://reviewboard.asterisk.org/r/3615/
+
+2014-06-16 08:52 +0000 [r416336]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: We have faced
+	  situation when using CDR and CEL by sqlite3 modules. With system
+	  having high load (~100 concurrent calls created by sipp) we found
+	  many cdr and cel records missed. There is special finction in
+	  sqlite3, that make able to fix this situation -
+	  sqlite3_wait_timeout, that also can replace awful code
+	  cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be
+	  used for aastdb and res_config_sqlite3 to avoid missed writes to
+	  sqlite db. #ASTERISK-23766 #close Reported by: Igor Goncharovsky
+	  Review: https://reviewboard.asterisk.org/r/3559/
+
+2014-06-15 21:16 +0000 [r416251]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This patch
+	  reverts r416150. When the comparison between mohclass->name and
+	  state->class->name is made, you are not guaranteed that (a)
+	  state->class is non-NULL or that state or state->class are in a
+	  safe state. Crashes caught by the bridges/transfer_capabilities
+	  test.
+
+2014-06-13 13:03 +0000 [r416150]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_musiconhold.c: MoH: Don't restart stream on repeated
+	  start calls Currently, music on hold will stop and then start
+	  again from the beginning if ast_moh_start() is called multiple
+	  times. This can happen if a call is put on hold repeatedly (the
+	  channel receives multiple HOLD control frames) and can be
+	  triggered from ARI by starting MoH on a channel multiple times.
+	  This is fairly jarring/annoying to users. This change prevents
+	  MoH from being restarted if the requested music class is the same
+	  as the one currently playing. Review:
+	  https://reviewboard.asterisk.org/r/3615/
+
+2014-06-13 04:58 +0000 [r416066]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+	  include/asterisk/tcptls.h: AST-2014-007: Fix of fix to allow AMI
+	  and SIP TCP to send messages. ASTERISK-23673 #close Reported by:
+	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/
+
+2014-06-12 21:15 +0000 [r415998]  Rusty Newton <rnewton at digium.com>
+
+	* main/pbx.c: main/pbx - documentation - enhance 'core show hints'
+	  and 'core show hint' help text Adds descriptive help text to
+	  'core show hints' and 'core show hint'. The text describes the
+	  various columns for the sake of clarity. ASTERISK-23764 Review:
+	  https://reviewboard.asterisk.org/r/3610/
+
+2014-06-12 17:16 +0000 [r415908]  Corey Farrell <git at cfware.com>
+
+	* channels/sip/sdp_crypto.c: chan_sip: DEBUG messages in
+	  sdp_crypto.c display despite a DEBUG level of zero Change debug
+	  level for messages in sdp_crypto.c from zero to one. This ensures
+	  the messages are not displayed when debugging is disabled. Change
+	  does not apply to 12+ as it was already fixed in those versions.
+	  ASTERISK-23246 #close Reported by: Rusty Newton Review:
+	  https://reviewboard.asterisk.org/r/3605/
+
+2014-06-12 16:05 +0000 [r415841]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/tcptls.h, configs/http.conf.sample,
+	  include/asterisk/utils.h, main/tcptls.c, main/manager.c,
+	  channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c:
+	  AST-2014-007: Fix DOS by consuming the number of allowed HTTP
+	  connections. Simply establishing a TCP connection and never
+	  sending anything to the configured HTTP port in http.conf will
+	  tie up a HTTP connection. Since there is a maximum number of open
+	  HTTP sessions allowed at a time you can block legitimate
+	  connections. A similar problem exists if a HTTP request is
+	  started but never finished. * Added http.conf session_inactivity
+	  timer option to close HTTP connections that aren't doing
+	  anything. Defaults to 30000 ms. * Removed the undocumented
+	  manager.conf block-sockets option. It interferes with TCP/TLS
+	  inactivity timeouts. * AMI and SIP TLS connections now have
+	  better authentication timeout protection. Though I didn't remove
+	  the bizzare TLS timeout polling code from chan_sip. * chan_sip
+	  can now handle SSL certificate renegotiations in the middle of a
+	  session. It couldn't do that before because the socket was
+	  non-blocking and the SSL calls were not restarted as documented
+	  by the OpenSSL documentation. * Fixed an off nominal leak of the
+	  ssl struct in handle_tcptls_connection() if the FILE stream
+	  failed to open and the SSL certificate negotiations failed. The
+	  patch creates a custom FILE stream handler to give the created
+	  FILE streams inactivity timeout and timeout after a specific
+	  moment in time capability. This approach eliminates the need for
+	  code using the FILE stream to be redesigned to deal with the
+	  timeouts. This patch indirectly fixes most of ASTERISK-18345 by

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