[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/.lastclean (with props)
tags/11.12.0-rc1/.version (with props)
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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