[svn-commits] bebuild: tag 11.12.0-rc1 r420807 - /tags/11.12.0-rc1/

SVN commits to the Digium repositories svn-commits at lists.digium.com
Mon Aug 11 13:50:38 CDT 2014


Author: bebuild
Date: Mon Aug 11 13:50:34 2014
New Revision: 420807

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=420807
Log:
Importing files for 11.12.0-rc1 release.

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    tags/11.12.0-rc1/ChangeLog   (with props)

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+2014-08-11  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.12.0-rc1 Released.
+
+2014-08-11 10:36 +0000 [r420655-420715]  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/ ........ Merged
+	  revisions 420680 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode.
+	  ........ Merged revisions 420654 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-07 21:37 +0000 [r420435]  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/ ........ Merged
+	  revisions 420434 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-06 16:08 +0000 [r420147]  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/ ........ Merged
+	  revisions 420146 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-05 18:23 +0000 [r420054]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/format.c: format.c: Add reason comments for the format_list
+	  ordering.
+
+2014-08-04 19:44 +0000 [r419943]  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 ........ Merged revisions
+	  419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-28 18:34 +0000 [r419685]  Richard Mudgett <rmudgett at digium.com>
+
+	* funcs/func_jitterbuffer.c, 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/ Audit of v11 usage of
+	  ast_channel_datastore_remove() for datastore memory leaks. *
+	  Fixed leak in func_jitterbuffer. Review:
+	  https://reviewboard.asterisk.org/r/3860/ ........ Merged
+	  revisions 419684 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-25 23:13 +0000 [r419631]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/features.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/ ........ Merged
+	  revisions 419630 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-24 17:56 +0000 [r419441]  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/
+	  ........ Merged revisions 419440 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-24 16:49 +0000 [r419375]  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) ........
+	  Merged revisions 419374 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-23 13:21 +0000 [r419284]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* apps/app_voicemail.c: app_voicemail: use a consistent generator
+	  string When updating voicemail.conf when a user changes their
+	  pin, change the generator string to be the same as the module
+	  name when reading so that the same config_hook will be called.
+	  Review: https://reviewboard.asterisk.org/r/3837/
+
+2014-07-22 14:00 +0000 [r419162]  Kinsey Moore <kmoore at digium.com>
+
+	* tests/test_voicemail_api.c, tests/test_aoc.c,
+	  tests/test_astobj2.c, tests/test_config.c,
+	  addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
+	  addons/chan_ooh323.c, tests/test_astobj2_thrash.c, /,
+	  apps/app_meetme.c, tests/test_abstract_jb.c, tests/test_logger.c,
+	  tests/test_event.c, tests/test_format_api.c,
+	  tests/test_hashtab_thrash.c, res/res_jabber.c: Fix more dev-mode
+	  build issues ........ Merged revisions 419129 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-15 22:05 +0000 [r418713]  Matthew Jordan <mjordan at digium.com>
+
+	* main/manager.c: manager: Return ActionID on nominal responses to
+	  PresenceState action When the PresenceState action is executed,
+	  the nominal path fails to include the ActionID in the successful
+	  response. This patch adds a call to astman_start_ack, which
+	  guarantees that an ActionID (if provided) will be sent back to
+	  the AMI client. Review: https://reviewboard.asterisk.org/r/3776/
+	  ASTERISK-23985 #close
+
+2014-07-15 17:32 +0000 [r418649]  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/ ........ Merged
+	  revisions 418641 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-13 21:51 +0000 [r418465-418505]  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/ ........ Merged
+	  revisions 418504 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* apps/app_skel.c: Fix minor reference leaks in app_skel and
+	  TEST_FRAMEWORK * Cleanup games object in app_skel. * Cleanup
+	  stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review:
+	  https://reviewboard.asterisk.org/r/3757/
+
+2014-07-11 14:23 +0000 [r418366]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/config.c: config: inform config hook of change when writing
+	  file When updated configuration is written back to the conf file
+	  - for example when a user changes their voicemail pin, make sure
+	  that any config hook that wants to know of changes is informed.
+	  Review: https://reviewboard.asterisk.org/r/3708/
+
+2014-07-10 15:35 +0000 [r418323]  Matthew Jordan <mjordan at digium.com>
+
+	* include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert
+	  indentation to tabs This is a whitespace only change.
+
+2014-07-10 01:42 +0000 [r418262]  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. ........ Merged revisions 418261 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-10  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.11.0 Released.
+
+2014-07-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.11.0-rc1 Released.
+
+2014-07-03 21:48 +0000 [r417957]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.h, channels/chan_dahdi.c,
+	  configs/chan_dahdi.conf.sample, /, UPGRADE.txt,
+	  channels/sig_pri.c: 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/ ........ Merged
+	  revisions 417956 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-03 11:24 +0000 [r417798]  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) ........ Merged revisions 417797 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-30 19:42 +0000 [r417677]  Joshua Colp <jcolp at digium.com>
+
+	* channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+	  main/rtp_engine.c, channels/chan_sip.c, UPGRADE.txt,
+	  configs/sip.conf.sample, include/asterisk/rtp_engine.h:
+	  res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS
+	  negotiation on RTCP. This change fixes up DTLS support in
+	  res_rtp_asterisk so it can accept and provide a SHA-256
+	  fingerprint, so it occurs on RTCP, and so it occurs after ICE
+	  negotiation completes. Configuration options to chan_sip have
+	  also been added to allow behavior to be tweaked (such as forcing
+	  the AVP type media transports in SDP). ASTERISK-22961 #close
+	  Reported by: Jay Jideliov Review:
+	  https://reviewboard.asterisk.org/r/3679/
+
+2014-06-30 03:23 +0000 [r417588]  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.
+	  Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916
+	  #close Reported by: Alexander Traud patches:
+	  sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud
+	  (License 6520) ........ Merged revisions 417587 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-27 19:26 +0000 [r417481-417505]  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/ ........ Merged
+	  revisions 417500 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, 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/ ........ Merged
+	  revisions 417480 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-26 18:25 +0000 [r417310-417419]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_http_websocket.exports.in: res_http_websocket: Export
+	  symbol for ast_websocket_set_timeout Thanks to Sean Bright for
+	  pointing out that this was missed in #asterisk-dev.
+
+	* 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) ........ Merged revisions 417318
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* UPGRADE.txt, configs/sip.conf.sample, res/res_http_websocket.c,
+	  channels/sip/include/sip.h, channels/chan_sip.c,
+	  include/asterisk/http_websocket.h: res_http_websocket: Close
+	  websocket correctly and use careful fwrite When a client takes a
+	  long time to process information received from Asterisk, a write
+	  operation using fwrite may fail to write all information. This
+	  causes the underlying file stream to be in an unknown state, such
+	  that the socket must be disconnected. Unfortunately, there are
+	  two problems with this in Asterisk's existing websocket code: 1.
+	  Periodically, during the read loop, Asterisk must write to the
+	  connected websocket to respond to pings. As such, Asterisk
+	  maintains a reference to the session during the loop. When
+	  ast_http_websocket_write fails, it may cause the session to
+	  decrement its ref count, but this in and of itself does not break
+	  the read loop. The read loop's write, on the other hand, does not
+	  break the loop if it fails. This causes the socket to get in a
+	  'stuck' state, preventing the client from reconnecting to the
+	  server. 2. More importantly, however, is that the fwrite in
+	  ast_http_websocket_write fails with a large volume of data when
+	  the client takes awhile to process the information. When it does
+	  fail, it fails writing only a portion of the bytes. With some
+	  debugging, it was shown that this was failing in a similar
+	  fashion to ASTERISK-12767. Switching this over to
+	  ast_careful_fwrite with a long enough timeout solved the problem.
+	  ASTERISK-23917 #close Reported by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3624/
+
+2014-06-26 10:04 +0000 [r417249]  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) ........ Merged revisions 417248 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-23 18:49 +0000 [r417141]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Return the length of
+	  data written when sending via ICE instead of 0. ASTERISK-23834
+	  #close Reported by: Richard Kenner
+
+2014-06-23 14:35 +0000 [r417077]  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 Review: https://reviewboard.asterisk.org/r/3622/
+
+2014-06-22 20:52 +0000 [r417017]  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/ ........
+	  Merged revisions 417016 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-20 23:14 +0000 [r416870-416930]  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/ ........ Merged
+	  revisions 416929 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* 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 19:34 +0000 [r416733]  Kinsey Moore <kmoore at digium.com>
+
+	* main/bridging.c, /, channels/sip/reqresp_parser.c, main/logger.c,
+	  main/test.c: Fix build warnings with TEST_FRAMEWORK enabled
+	  ........ Merged revisions 416732 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-19 16:02 +0000 [r416581-416668]  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)'. ASTERISK-23844 #close Reported by: rnewton Tested by:
+	  gtjoseph Review: https://reviewboard.asterisk.org/r/3627/
+
+2014-06-17 18:40 +0000 [r416501]  Mark Michelson <mmichelson at digium.com>
+
+	* /, funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to
+	  set inheritable channel variables. ........ Merged revisions
+	  416500 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-17 16:21 +0000 [r416440]  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/ ........ Merged
+	  revisions 416439 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-16 09:00 +0000 [r416337]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* cel/cel_sqlite3_custom.c, main/db.c, res/res_config_sqlite3.c,
+	  cdr/cdr_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/ ........ Merged
+	  revisions 416336 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-15 21:17 +0000 [r416252]  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. ........ Merged revisions 416251 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-13 13:08 +0000 [r416151]  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/ ........ Merged
+	  revisions 416150 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-13 05:06 +0000 [r416067]  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/ ........ Merged
+	  revisions 416066 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 21:16 +0000 [r415999]  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/ ........ Merged
+	  revisions 415998 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 17:20 +0000 [r415915]  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/ ........ Merged
+	  revisions 415908 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 16:22 +0000 [r415854]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_http_websocket.c, 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,
+	  include/asterisk/tcptls.h: 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 fixing the usage of the SSL_read/SSL_write operations.
+	  ASTERISK-23673 #close Reported by: Richard Mudgett ........
+	  Merged revisions 415841 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 15:42 +0000 [r415837]  Jonathan Rose <jrose at digium.com>
+
+	* UPGRADE.txt: Correct UPGRADE.txt notes in r415825 The change was
+	  marked against the wrong version of Asterisk. My apologies.
+
+2014-06-12 15:40 +0000 [r415835]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* /, apps/app_queue.c: app_queue: delayed state can cause early
+	  leavewhenempty ringing In app_queue, device state changes arrive
+	  in event messages and update the queue member status value. That
+	  value is checked in get_member_status() to decide that the caller
+	  should leave when there are no available members. Although event
+	  messages can be delayed by other activity, there is no adverse
+	  affect by lagged status except in one specific case: there is
+	  only one available member, it was just rung, and leavewhenempty
+	  is enabled set for ringing members. This change adds a direct
+	  check of the device state only under this condition where the
+	  caller may be dropped incorrectly, resolving this issue without
+	  affecting performance of app_queue normally. AST-1248 #close
+	  Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
+	  Thomas Arimont ........ Merged revisions 415833 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-12 15:22 +0000 [r415825]  Jonathan Rose <jrose at digium.com>
+
+	* UPGRADE.txt, apps/app_mixmonitor.c: MixMonitor: Add class
+	  authorization requirements to MixMonitor AMI commands MixMonitor
+	  AMI commands StartMixMonitor and StopMixMonitor lacked class
+	  authorization. StopMixMonitor now requires that the manager user
+	  either have the call or system class authorization.
+	  StartMixMonitor is a slightly larger issue since it can execute
+	  shell commands if the right arguments are passed into it, and we
+	  consider this a permission escalation. A security release will be
+	  issued for problem this shortly. ASTERISK-23609 #close Reported
+	  by: Corey Farrell
+
+2014-06-11 22:44 +0000 [r415728]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/format.c: format.c: Fix misuse of hash container function.
+	  The supplied hash function to a container must be idempotent
+	  given the object's key value to figure out which container bucket
+	  the object belongs in. Returning a random number or the current
+	  container count is not idempotent. The "computed hash" value
+	  doesn't help find the object later in those cases. * Fixed the
+	  format_list container to actually be a list since that is how the
+	  container is used. Conceptually, if more than 283 formats were
+	  added to the format_list then odd things may have happened before
+	  the fix.
+
+2014-06-10 09:13 +0000 [r415599]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: chan_ooh323: fix loading module failure if
+	  there no accessible h323_log or ooh323 config file change return
+	  1 to return AST_MODULE_LOAD_FAILURE on module load routine few
+	  cosmetic changes ASTERISK-23814 #close (closes issue
+	  ASTERISK-23814) Reported by: Igor Goncharovsky Patches:
+	  ASTERISK-23814-ast11.patch
+
+2014-06-09 11:57 +0000 [r415522]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* contrib/scripts/safe_asterisk, /: safe_asterisk: Cleanup
+	  additions to r415132. Replaced a stray echo that should've been a
+	  message call in safe_asterisk. I'm using the contents of the old
+	  message inside the if $NOTIFY so peoples log parsing scripts
+	  won't get confused by new messages. I'll clean that up in trunk.
+	  (Note that a 'make install' still won't overwrite your old
+	  safe_asterisk if it exists. See ASTERISK-21965.) ASTERISK-23492
+	  #close ........ Merged revisions 415521 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-09 03:47 +0000 [r415464]  Corey Farrell <git at cfware.com>
+
+	* main/autoservice.c, /: autoservice: stop thread on graceful
+	  shutdown This change adds thread shutdown to autoservice for
+	  graceful shutdowns only. ast_register_cleanup is backported to
+	  1.8 to allow this. The logger callid is also released on shutdown
+	  in 11+. ASTERISK-23827 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3594/ ........ Merged
+	  revisions 415463 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-06 21:27 +0000 [r415390]  Jonathan Rose <jrose at digium.com>
+

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