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.

Added:
    tags/11.12.0-rc1/.lastclean   (with props)
    tags/11.12.0-rc1/.version   (with props)
    tags/11.12.0-rc1/ChangeLog   (with props)

Added: tags/11.12.0-rc1/.lastclean
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.12.0-rc1/.lastclean?view=auto&rev=420807
==============================================================================
--- tags/11.12.0-rc1/.lastclean (added)
+++ tags/11.12.0-rc1/.lastclean Mon Aug 11 13:50:34 2014
@@ -1,0 +1,1 @@
+40

Propchange: tags/11.12.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.12.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.12.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.12.0-rc1/.version
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.12.0-rc1/.version?view=auto&rev=420807
==============================================================================
--- tags/11.12.0-rc1/.version (added)
+++ tags/11.12.0-rc1/.version Mon Aug 11 13:50:34 2014
@@ -1,0 +1,1 @@
+11.12.0-rc1

Propchange: tags/11.12.0-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.12.0-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.12.0-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.12.0-rc1/ChangeLog
URL: 
http://svnview.digium.com/svn/asterisk/tags/11.12.0-rc1/ChangeLog?view=auto&rev=420807
==============================================================================
--- tags/11.12.0-rc1/ChangeLog (added)
+++ tags/11.12.0-rc1/ChangeLog Mon Aug 11 13:50:34 2014
@@ -1,0 +1,31081 @@
+2014-08-11  Asterisk Development Team <asteriskt...@digium.com>
+
+       * Asterisk 11.12.0-rc1 Released.
+
+2014-08-11 10:36 +0000 [r420655-420715]  Walter Doekes <walter+aster...@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 <rmudg...@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.jos...@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 <rmudg...@digium.com>
+
+       * main/format.c: format.c: Add reason comments for the format_list
+         ordering.
+
+2014-08-04 19:44 +0000 [r419943]  Rusty Newton <rnew...@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 <rmudg...@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 <rmudg...@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 <g...@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 <jpar...@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 <sgriepent...@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 <kmo...@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 <mjor...@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 <jr...@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 <g...@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 <sgriepent...@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 <mjor...@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 <rmudg...@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 <asteriskt...@digium.com>
+
+       * Asterisk 11.11.0 Released.
+
+2014-07-08  Asterisk Development Team <asteriskt...@digium.com>
+
+       * Asterisk 11.11.0-rc1 Released.
+
+2014-07-03 21:48 +0000 [r417957]  Richard Mudgett <rmudg...@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 <mjor...@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 <jc...@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 <mjor...@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 <g...@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 <mjor...@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 <g...@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 <jc...@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 <rnew...@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.jos...@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.jos...@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 <kmo...@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.jos...@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 <mmichel...@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 <kmo...@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.goncharov...@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 <mjor...@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 <kmo...@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 <rmudg...@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 <rnew...@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 <g...@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 <rmudg...@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 <jr...@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 <sgriepent...@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 <jr...@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 <rmudg...@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 <m...@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+aster...@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 <g...@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 <jr...@digium.com>
+

[... 30385 lines stripped ...]

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

svn-commits mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/svn-commits

Reply via email to