[svn-commits] bebuild: tag 10.12.0-rc1 r377518 - /tags/10.12.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Dec 9 19:52:00 CST 2012
Author: bebuild
Date: Sun Dec 9 19:51:58 2012
New Revision: 377518
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=377518
Log:
Importing files for 10.12.0-rc1 release.
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tags/10.12.0-rc1/.lastclean (with props)
tags/10.12.0-rc1/.version (with props)
tags/10.12.0-rc1/ChangeLog (with props)
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+2012-12-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.12.0-rc1 Released.
+
+2012-12-10 01:39 +0000 [r377504-377510] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/xmldoc.c, /: Improve documentation by making all of the
+ colors used readable, no matter what the background color is.
+ Dark blue on a black background is unreadable, as is yellow on a
+ light background. This patch turns on the bright attribute for
+ colors when on a dark background and turns *off* the bright
+ attribute when the -W command line option is used (indicating a
+ _light_ background). This ensures that text is readable in both
+ cases. Patch by: tilghman Review:
+ https://reviewboard.asterisk.org/r/2224 ........ Merged revisions
+ 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, addons/cdr_mysql.c: Remove some dead code and additionally
+ handle a case that wasn't handled. ........ Merged revisions
+ 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-08 00:29 +0000 [r377399-377432] Richard Mudgett <rmudgett at digium.com>
+
+ * contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP
+ allow/disallow in MySQL contrib script. Using the contrib
+ sippeers.sql script to create the sippeers MySQL table would
+ result in being unable to place calls if you set the disallow
+ value to all. (closes issue ASTERISK-20756) Reported by: Andre
+ Luis Patches: sippeers.patch patch uploaded by Andre Luis
+ ........ Merged revisions 377431 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit
+ allocation dumps. ........ Merged revisions 377398 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-07 21:58 +0000 [r377382] Kinsey Moore <kmoore at digium.com>
+
+ * codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder
+ show" CLI command. In r306010 "Asterisk media architecture
+ conversion - no more format bitfields", the logic for
+ incrementing encoders and decoders when opening transcoder
+ channels was changed without making the corresponding change when
+ decrementing encoder / decoder channels. The result being that
+ when a channel was destroyed, codec_dahdi couldn't properly tell
+ if it was an encoder or decoder, and the default case is to
+ assume it was a decoder. This could result in negative numbers
+ for decoders in use like in: VOIP6*CLI> transcoder show 2/-2
+ encoders/decoders of 92 channels are in use. (closes issue
+ ASTERISK-19921) Patch-by: Shaun Ruffell
+
+2012-12-06 23:56 +0000 [r377354] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_config_parser.c, apps/app_confbridge.c:
+ confbridge: Fix some resource leaks on conference teardown. *
+ Made destroy_conference_bridge() destroy a missed ast_mutex_t and
+ ast_cond_t. * Made join_conference_bridge() init the
+ ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can
+ destroy them unconditionally. * Made join_conference_bridge()
+ abort if the new conference could not be added to the conferences
+ container. * Made leave_conference() discard any post-join
+ actions if join_conference_bridge() had to abort early. * Made
+ the join_conference_bridge() diagnostic messages better describe
+ what happened. * Renamed leave_conference_bridge() to
+ leave_conference() and made it only take a conference user
+ pointer. The conference pointer was redundant. * Made
+ conf_bridge_profile_copy() use struct copy instead of memcpy(). *
+ No need to lock the conference in start_conf_record_thread()
+ since all of the callers already have it locked.
+
+2012-12-05 16:57 +0000 [r377261] Jonathan Rose <jrose at digium.com>
+
+ * res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc
+ on an already dealloced session When srtp_create fails, the
+ session may be dealloced or just not alloced. At the same time
+ though, the session pointer might not be set to NULL in this
+ process and attempting to srtp_dealloc it again will cause a
+ segfault. This patch checks for failure of srtp_create and sets
+ the session pointer to NULL if it fails. (closes issue
+ ASTERISK-20499) Reported by: tootai Review:
+ https://reviewboard.asterisk.org/r/2228/ ........ Merged
+ revisions 377256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 16:49 +0000 [r377258] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Fix a SIP request memory leak with TLS
+ connections. During the TLS re-work in chan_sip some TLS specific
+ code was moved into a separate function. This function operates
+ on a copy of the incoming SIP request. This copy was never
+ deinitialized causing a memory leak for each request processed.
+ This function is now given a SIP request structure which it can
+ use to copy the incoming request into. This reduces the amount of
+ memory allocations done since the internal allocated components
+ are reused between packets and also ensures the SIP request
+ structure is deinitialized when the TLS connection is torn down.
+ (closes issue ASTERISK-20763) Reported by: deti ........ Merged
+ revisions 377257 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-05 02:09 +0000 [r377038-377241] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format.c: * Fix registering core show codecs/codec CLI
+ commands twice. * Fix registering atexit format_attr_shutdown()
+ more than once.
+
+ * apps/confbridge/conf_config_parser.c: confbridge: Fix several
+ small issues. * Made func_confbridge_helper() allow an empty
+ value when setting options. You previously could not
+ Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the
+ dialplan. * Made func_confbridge_helper() handle its datastore
+ better if multiple threads attempt to set the first CONFBRIDGE
+ option value on the channel. * Made the func_confbridge_helper()
+ only output one diagnostic message concerning the option. * Made
+ the bridge video_mode able to repeatedly change in the config
+ file and CONFBRIDGE dialplan function. The video_mode option
+ values are an enum and not independent of each other. * Made
+ handle_cli_confbridge_show_bridge_profile() better handle the
+ video_mode option. * Simplified datastore handling code in
+ conf_find_user_profile() and conf_find_bridge_profile(). * Made
+ parse_bridge(), parse_user(), and parse_menu() use var->file
+ instead of CONFBRIDGE_CONFIG because the var could have been from
+ an include file. (closes issue ASTERISK-20655) Reported by:
+ Birger "WIMPy" Harzenetter
+
+ * apps/app_confbridge.c: confbridge: Update online XML
+ documentation.
+
+ * /, main/asterisk.c: Cleanup ast_run_atexits() atexits list. *
+ Convert atexits list to a mutex instead of a rd/wr lock. The lock
+ is only write locked. * Move CLI verbose Asterisk ending message
+ to where AMI message is output in really_quit() to avoid further
+ surprises about using stuff already shutdown. (issue
+ ASTERISK-20649) Reported by: Corey Farrell ........ Merged
+ revisions 377165 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/_private.h, main/stdtime/localtime.c,
+ main/asterisk.c: Cleanup core main on exit. * Cleanup time zones
+ on exit. * Make exit clean/unclean report consistent for AMI and
+ CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell core-cleanup-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377135 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/config.c: Cleanup config cache on exit. (issue
+ ASTERISK-20649) Reported by: Corey Farrell Patches:
+ config-cleanup-all.patch (license #5909) patch uploaded by Corey
+ Farrell ........ Merged revisions 377104 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cli.c, /: Cleanup CLI resources on exit and CLI command
+ registration errors. (issue ASTERISK-20649) Reported by: Corey
+ Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch
+ uploaded by Corey Farrell cli-leaks-11-trunk.patch (license
+ #5909) patch uploaded by Corey Farrell Modified ........ Merged
+ revisions 377073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cdr.c, /: Cleanup CDR resources on exit. * Simplify
+ do_reload() return handling since it never returned anything
+ other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by
+ Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch
+ uploaded by Corey Farrell Modified ........ Merged revisions
+ 377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/ccss.c: Fix CCSS CLI commands and logger level not
+ unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell
+ Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by
+ Corey Farrell ........ Merged revisions 377037 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 21:33 +0000 [r376951] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending
+ RELEASE_COMPLETE in response to SETUP. Fix sending a
+ RELEASE_COMPLETE in response to a SETUP if chan_misdn does not
+ have a B channel available to assign to the call. (closes issue
+ ABE-2869) Reported by: Guenther Kelleter Patches:
+ setup-reject_2.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified ........ Merged revision 376949 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 376950 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 17:06 +0000 [r376920] Sean Bright <sean at malleable.com>
+
+ * /, funcs/func_volume.c: Minor spelling fix to the VOLUME
+ documentation. ........ Merged revisions 376919 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-30 16:23 +0000 [r376916] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Fix potential crashes during SIP attended
+ transfers. The principal behind this patch is simple. During a
+ transfer, we manipulate channels that are owned by a separate
+ thread than the one we currently are running in, so it makes
+ sense that we need to grab a reference to the channels so that
+ they cannot disappear out from under us. In the wild, crashes
+ were sometimes seen when the transferring party would hang up the
+ call before the transfer target answered the call. The most
+ common place to see the crash occur was when attempting to send a
+ connected line update to the transferer channel. (closes issue
+ ASTERISK-20226) Reported by Jared Smith Patches:
+ ASTERISK-20226.patch uploaded by Mark Michelson (License #5049)
+ Tested by: Jared Smith ........ Merged revisions 376901 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 22:58 +0000 [r376865-376869] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in
+ local_devicestate(). Regression introduced by ASTERISK-20390 fix.
+ (closes issue ASTERISK-20769) Reported by: rmudgett Tested by:
+ rmudgett ........ Merged revisions 376868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724)
+ ........ Merged revisions 376864 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 21:51 +0000 [r376835] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, channels/chan_sip.c: Improve Code Readability And Fix Setting
+ natdetected Flag For 1.8, 10, 11 and trunk we are are improving
+ the code readability. For 11 and trunk, auto nat detection was
+ added. The natdetected flag was being set to 1 when the host
+ address in the VIA header did not specifiy a port. This patch
+ fixes this by setting the port on the temporary sock address used
+ to SIP_STANDARD_PORT in order for the sock address comparison to
+ work properly. (closes issue ASTERISK-20724) Reported by: Michael
+ L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2206/ ........ Merged
+ revisions 376834 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-29 00:45 +0000 [r376759-376789] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG atexit
+ unreleased malloc memory summary. * Adds the following CLI
+ commands to control MALLOC_DEBUG reporting of unreleased malloc
+ memory when Asterisk is shut down. memory atexit list on memory
+ atexit list off memory atexit summary byline memory atexit
+ summary byfunc memory atexit summary byfile memory atexit summary
+ off * Made check all remaining allocated region blocks atexit for
+ fence violations. * Increased the allocated region hash table
+ size by about three times. It still isn't large enough
+ considering the number of malloced blocks Asterisk uses. * Made
+ CLI "memory show allocations anomalies" use
+ regions_check_all_fences(). Review:
+ https://reviewboard.asterisk.org/r/2196/ ........ Merged
+ revisions 376788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI
+ "memory show allocations" misspelling of anomalies option. The
+ command will still accept the original misspelling. *
+ Miscellaneous tweaks to CLI "memory show allocations" command
+ output format. * Made CLI "memory show summary" summarize by line
+ number instead of by function if a filename is given. * Made CLI
+ "memory show summary" sort its output by filename or
+ function-name/line-number depending upon request. * Miscellaneous
+ tweaks to CLI "memory show summary" command output format.
+ ........ Merged revisions 376758 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-28 16:30 +0000 [r376726] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: manager: Make challenge work with
+ allowmultiplelogin=no Prior to this patch, challenge would yield
+ a multiple logins error if used without providing the username
+ (which isn't really supposed to be an argument to challenge) if
+ allowmultiplelogin was set to no because allowmultiplelogin finds
+ a user with a zero length login name. This check is simply
+ disabled for the challenge action when the username is empty by
+ this patch. (closes issue ASTERISK-20677) Reported by: Vladimir
+ Patches: challenge_action_nomultiplelogin.diff uploaded by
+ Jonathan Rose (license 6182) ........ Merged revisions 376725
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-27 23:58 +0000 [r376628-376689] Richard Mudgett <rmudgett at digium.com>
+
+ * main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-'
+ char. The '-' char is supposed to be ignored by the dialplan
+ extension matching. Unfortunately, it's treatment is not handled
+ consistently throughout the extension matching code. * Made the
+ old exten matching code consistently ignore '-' chars. * Made the
+ old exten matching code consistently handle case in the matching.
+ * Made ignore empty character sets. * Fixed ast_extension_cmp()
+ to return -1, 0, or 1 as documented. The only user of it in
+ pbx_lua.c was testing for -1. It was originally returning the
+ strcmp() value for less than which is not usually going to be -1.
+ * Fix character set sorting if the sets have the same number of
+ characters and start with the same character. Character set [0-9]
+ now sorts before [02-9a] as originally intended. * Updated some
+ extension label and priority already in use warnings to also
+ indicate if the extension is aliased. (closes issue
+ ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy"
+ Harzenetter Tested by: rmudgett Review:
+ https://reviewboard.asterisk.org/r/2201/ ........ Merged
+ revisions 376688 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * addons/res_config_mysql.c, /, apps/app_celgenuserevent.c,
+ pbx/pbx_dundi.c: Remove unnecessary channel module references. *
+ Removed call to ast_module_user_hangup_all() in
+ res_config_mysql.c since it is effectively a noop. No channels
+ can attach a reference to that module. * Removed call to
+ ast_module_user_hangup_all() in app_celgenuserevent.c. The caller
+ of unload_module() has already called it. * Removed redundant
+ channel module references in pbx_dundi.c. The registered dialplan
+ function callback dispatchers for the read/read2/write callbacks
+ already reference the module before calling. * pbx_dundi: Moved
+ unregistering CLI commands, DUNDi switch, and dialplan functions
+ to the first thing the unload_module() does. This will reduce the
+ chance of new channels using DUNDi services while the module is
+ being torn down. ........ Merged revisions 376657 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler
+ and use better names. * Update doxygen of AST_LIST_REMOVE().
+ ........ Merged revisions 376627 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-22 23:56 +0000 [r376587] Matthew Jordan <mjordan at digium.com>
+
+ * main/lock.c, /, main/logger.c, include/asterisk/lock.h:
+ Re-initialize logmsgs mutex upon logger initialization to prevent
+ lock errors Similar to the patch that moved the fork earlier in
+ the startup sequence to prevent mutex errors in the recursive
+ mutex surrounding the read/write thread registration lock, this
+ patch re-initializes the logmsgs mutex. Part of the start up
+ sequence before forking the process into the background includes
+ reading asterisk.conf; this has to occur prior to the call to
+ daemon in order to read startup parameters. When reading in a
+ conf file, log statements can be generated. Since this can't be
+ avoided, the mutex instead is re-initialized to ensure a reset of
+ any thread tracking information. This patch also includes some
+ additional debugging to catch errors when locking or unlocking
+ the recursive mutex that surrounds locks when the DEBUG_THREADS
+ build option is enabled. DO_CRASH or THREAD_CRASH will cause an
+ abort() if a mutex error is detected. (issue ASTERISK-19463)
+ Reported by: mjordan Tesetd by: mjordan ........ Merged revisions
+ 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-20 17:01 +0000 [r376522] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: Add "Require:
+ timer" to 200 OK responses when appropriate. The method by which
+ the Require header is added to 200 responses is inspired by the
+ method that Olle Johansson uses in his darjeeling-prack branch.
+ (closes issue ASTERISK-20570) Reported by Matt Jordan, at the
+ behest of Olle Johansson Review:
+ https://reviewboard.asterisk.org/r/2172 ........ Merged revisions
+ 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-19 19:44 +0000 [r376470] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c, main/security_events.c,
+ main/indications.c: Fix most leftover non-opaque ast_str uses.
+ Instead of calling str->str, one should use ast_str_buffer(str).
+ Same goes for str->used as ast_str_strlen(str) and str->len as
+ ast_str_size(str). Review:
+ https://reviewboard.asterisk.org/r/2198 ........ Merged revisions
+ 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-18 20:18 +0000 [r376414-376431] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/utils.c, main/stdtime/localtime.c, main/asterisk.c:
+ Reorder startup sequence to prevent lockups when process is sent
+ to background Although it is very rare and timing dependent, the
+ potential exists for the call to 'daemon' to cause what appears
+ to be a deadlock in Asterisk during startup. This can occur when
+ a recursive mutex is obtained prior to the daemon call executing.
+ Since daemon uses fork to send the process into the background,
+ any threading primitives are unsafe to re-use after the call.
+ Implementations of pthread recursive mutexes are highly likely to
+ store the thread identifier of the thread that previously
+ obtained the mutex. If the mutex was locked prior to the fork, a
+ subsequent unlock operation will potentially fail as the thread
+ identifier is no longer valid. Since the mutex is still locked,
+ all subsequent attempts to grab the mutex by other threads will
+ block. This behavior exhibited itself most often when
+ DEBUG_THREADS was enabled, as this compile time option surrounds
+ the mutexes in Asterisk with another recursive mutex that
+ protects the storage of thread related information. This made it
+ much more likely that a recursive mutex would be obtained prior
+ to daemon and unlocked after the call. This patch does the
+ following: a) It backports a patch from Asterisk 11 that prevents
+ the spawning of the localtime monitoring thread. This thread is
+ now spawned after Asterisk has fully booted. b) It re-orders the
+ startup sequence to call daemon earlier during Asterisk startup.
+ This limits the potential of threading primitives being accessed
+ by initialization calls before daemon is called. c) It removes
+ calls to ast_verbose/ast_log/etc. prior to daemon being called.
+ Developers should send error messages directly to stderr prior to
+ daemon, as calls to ast_log may access recursive mutexes that
+ store thread related information. d) It reorganizes when thread
+ local storage is created for storing lock information during the
+ creation of threads. Prior to this patch, the read/write lock
+ protecting the list of threads in ast_register_thread would
+ utilize the lock in the thread local storage prior to it being
+ initialized; this patch prevents that. On a very related note,
+ this patch will *greatly* improve the stability of the Asterisk
+ Test Suite. Review: https://reviewboard.asterisk.org/r/2197
+ (closes issue ASTERISK-19463) Reported by: mjordan Tested by:
+ mjordan ........ Merged revisions 376428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/confbridge/conf_state.c: Add a test event that reports
+ changes in ConfBridge state This patch adds a test event to
+ ConfBridge that reports transitions between states in ConfBridge.
+ This is used by tests in the Asterisk Test Suite that verify
+ state changes based on the entering/leaving of conference
+ participants.
+
+2012-11-16 19:41 +0000 [r376390] Jonathan Rose <jrose at digium.com>
+
+ * /, res/res_monitor.c: monitor: prevent attempts to move/remove
+ recordings skipped with 'i' and 'o'. The i and o options for
+ monitor skip the input and output sides of a recording
+ respectively. This patch addresses a problem in those options
+ when monitor is called without specifying a specific filename
+ where monitor will try to move the recording that was skipped.
+ Since this usually doesn't exist when these options are used, it
+ would produce a warning when it does this in most cases, but it
+ is conceivable that there are use cases where this could result
+ in moving/removing a file unintentionally. (closes issue
+ ASTERISK-20641) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2190/ ........ Merged
+ revisions 376389 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-16 00:08 +0000 [r376315-376342] David M. Lee <dlee at digium.com>
+
+ * /, utils/extconf.c: Fixed extconf.c breakage introduced in
+ r376306. To quote wdoekes: > Note that I'm not confirming
+ legitimacy of having that file in tree at > all. Is anyone using
+ aelparse/conf2ael? ........ Merged revisions 376340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils/Makefile, tests/test_astobj2_thrash.c (added),
+ utils/utils.xml, /, utils/hashtest.c (removed),
+ tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed),
+ include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit
+ tests. Both hashtest and hashtest2 are manual testing apps that
+ thrash hash tables (hashtab and ao2 containers, respectively), by
+ spinning up several threads that randomly insert, delete, lookup
+ and iterate over the hash table. If the app doesn't crash, the
+ hash table probably passes the test. Those utils are not a part
+ of the typical Asterisk build, so they do not usually get
+ compiled. This all makes them less that useful. This patch
+ removes those manual test programs and replaces them with
+ Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It
+ also attempts to make the tests more deterministic. * Rather than
+ spinning up some number of threads that operate on the hash table
+ randomly, spin up four threads that concurrenly add, remove,
+ lookup and iterate over the hash table. * Each thread checks the
+ state of the hash table both during and after execution, and
+ indicates a test failure if things are not as expected. * Each
+ thread times out after 60 seconds to prevent deadlocking the unit
+ test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged
+ revisions 376306 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-15 22:55 +0000 [r376308] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_meetme.c: app_meetme: Fix channels lingering when
+ hung up under certain conditions Channels would get stuck and
+ MeetMe would repeatedly display an Unable to write frame to
+ channel error in the conf_run function if hung up during certain
+ sound prompts such as during user count announcements. This patch
+ fixes that by reintroducing a hangup check in the meetme's main
+ loop (also in conf_run). (closes issue ASTERISK-20486) Reported
+ by: Michael Cargile Review:
+ https://reviewboard.asterisk.org/r/2187/ Patches:
+ meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan
+ Rose (license 6182) ........ Merged revisions 376307 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-15 01:50 +0000 [r376263] Rusty Newton <rnewton at digium.com>
+
+ * apps/app_voicemail.c, /: Patch to play correct sound file when a
+ voicemail's urgent status is removed We were attempting to play
+ "vm-urgent-removed", which didn't exist. Now we play
+ "vm-marked-nonurgent" which exists and is the correct sound file.
+ Previous behavior was silence and a warning on the CLI. (issue
+ ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo
+ Takebe Tested by: Rusty Newton Patches: asterisk20280.patch
+ uploaded by Rusty Newton (license 5829) ........ Merged revisions
+ 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-14 19:50 +0000 [r376233] Richard Mudgett <rmudgett at digium.com>
+
+ * pbx/pbx_spool.c, /: Fix call files when astspooldir is relative.
+ Future dated call files are ignored when astspooldir is relative
+ to the current directory. The queue_file() assumed that the qdir
+ needed to be prepended if the given filename did not start with a
+ '/'. If astspooldir is relative it is not going to start from the
+ root directory obviously so it will not start with a '/'. The
+ filename used in queue_file() ultimately results in qdir
+ prepended multiple times. * Made queue_file() not prepend qdir if
+ the filename contains a '/'. (closes issue ASTERISK-20593)
+ Reported by: James Le Cuirot Patches:
+ 0004-Fix-future-call-files-from-relative-directories.patch
+ (license #6439) patch uploaded by James Le Cuirot ........ Merged
+ revisions 376232 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-13 18:20 +0000 [r376208] Brent Eagles <beagles at digium.com>
+
+ * main/channel.c, /: Patch to prevent stopping the active generator
+ when it is not the silence generator. This patch introduces an
+ internal helper function to safely check whether the current
+ generator is the one that is expected before deactivating it. The
+ current externally accessible ast_channel_stop_generator()
+ function has been modified to be implemented in terms of the new
+ function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad
+ ........ Merged revisions 376199 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-12 20:44 +0000 [r376167] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c, /: Properly check if the "Context" and "Extension"
+ headers are empty in a ShowDialPlan action. The code which
+ handles the ShowDialPlan action wrongly assumed that a non-NULL
+ return value from the function which retrieves headers from an
+ action indicates that the header has a value. This is incorrect
+ and the contents must be checked to see if they are blank.
+ (closes issue ASTERISK-20628) Reported by: jkroon Patches:
+ asterisk-showdialplan-incorrect-error.patch uploaded by jkroon
+ ........ Merged revisions 376166 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-12 20:15 +0000 [r376143] Michael L. Young <elgueromexicano at gmail.com>
+
+ * main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore
+ Problem When adding a dynamic hint, if an extension contains an
+ underscore no variable subsitution is being performed. This patch
+ changes from checking if the extension contains an underscore to
+ checking if the extension begins with an underscore. (closes
+ issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by:
+ Steven T. Wheeler, Michael L. Young Patches:
+ asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2188/ ........ Merged
+ revisions 376142 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-08 21:59 +0000 [r376088] Mark Michelson <mmichelson at digium.com>
+
+ * /, res/res_fax.c: Fix a "set but not used" warning on newer gccs.
+ Turns out the "helpful" setting of ms and res in this macro is
+ completely useless after the timeout antipattern fix. If you're a
+ new guy looking to write code, don't write a macro like this one.
+ ........ Merged revisions 376087 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-08 21:07 +0000 [r376030-376059] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_ss7.c: chan_dahdi/SS7: Made reject incoming call
+ for an in-alarm or blocked channel. If a SS7 call comes in
+ requesting a CIC that is in-alarm, the call is accepted and
+ connects if the extension exists in the dialplan. The call does
+ not have any audio. * Made release the call immediately with
+ circuit congestion cause. (closes issue ASTERISK-20204) Reported
+ by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license
+ #5621) patch uploaded by rmudgett ........ Merged revisions
+ 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * include/asterisk/utils.h, include/asterisk/astmm.h, /,
+ main/utils.c, main/astmm.c, main/asterisk.c: Add MALLOC_DEBUG
+ enhancements. * Makes malloc() behave like calloc(). It will
+ return a memory block filled with 0x55. A nonzero value. * Makes
+ free() fill the released memory block and boundary fence's with
+ 0xdeaddead. Any pointer use after free is going to have a pointer
+ pointing to 0xdeaddead. The 0xdeaddead pointer is usually an
+ invalid memory address so a crash is expected. * Puts the freed
+ memory block into a circular array so it is not reused
+ immediately. * When the circular array rotates out a memory block
+ to the heap it checks that the memory has not been altered from
+ 0xdeaddead. * Made the astmm_log message wording better. * Made
+ crash if the DO_CRASH menuselect option is enabled and something
+ is found. * Fixed a potential alignment issue on 64 bit systems.
+ struct ast_region.data[] should now be aligned correctly for all
+ platforms. * Extracted region_check_fences() from
+ __ast_free_region() and handle_memory_show(). * Updated
+ handle_memory_show() CLI usage help. Review:
+ https://reviewboard.asterisk.org/r/2182/ ........ Merged
+ revisions 376029 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-07 17:16 +0000 [r375995] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c,
+ main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c,
+ res/res_fax.c, apps/app_record.c, channels/chan_agent.c,
+ main/utils.c, include/asterisk/channel.h, apps/app_queue.c,
+ channels/sig_pri.c, channels/chan_iax2.c, main/channel.c,
+ channels/chan_dahdi.c, apps/app_waitforring.c,
+ channels/sig_analog.c: Multiple revisions 375993-375994 ........
+ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov
+ 2012) | 30 lines Fix misuses of timeouts throughout the code.
+ Prior to this change, a common method for determining if a
+ timeout was reached was to call a function such as
+ ast_waitfor_n() and inspect the out parameter that told how many
+ milliseconds were left, then use that as the input to
+ ast_waitfor_n() on the next go-around. The problem with this is
+ that in some cases, submillisecond timeouts can occur, resulting
+ in the out parameter not decreasing any. When this happens
+ thousands of times, the result is that the timeout takes much
+ longer than intended to be reached. As an example, I had a
+ situation where a 3 second timeout took multiple days to finally
+ end since most wakeups from ast_waitfor_n() were under a
+ millisecond. This patch seeks to fix this pattern throughout the
+ code. Now we log the time when an operation began and find the
+ difference in wall clock time between now and when the event
+ started. This means that sub-millisecond timeouts now cannot play
+ havoc when trying to determine if something has timed out. Part
+ of this fix also includes changing the function ast_waitfor() so
+ that it is possible for it to return less than zero when a
+ negative timeout is given to it. This makes it actually possible
+ to detect errors in ast_waitfor() when there is no timeout.
+ (closes issue ASTERISK-20414) reported by David M. Lee Review:
+ https://reviewboard.asterisk.org/r/2135/ ........ r375994 |
+ mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3
+ lines Remove some debugging that accidentally made it in the last
+ commit. ........ Merged revisions 375993-375994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-11-06 18:27 +0000 [r375965] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/features.h, main/channel.c, /, main/features.c,
+ include/asterisk/channel.h, .cleancount: Fix stuck DTMF when
+ bridge is broken. When a bridge is broken by an AMI Redirect
+ action or the ChannelRedirect application, an in progress DTMF
+ digit could be stuck sending forever. * Made simulate a DTMF end
+ event when a bridge is broken and a DTMF digit was in progress.
+ (closes issue ASTERISK-20492) Reported by: Jeremiah Gowdy
+ Patches: bridge_end_dtmf-v3.patch.txt (license #6358) patch
+ uploaded by Jeremiah Gowdy Modified to
+ jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch
+ (license #5621) patch uploaded by rmudgett Tested by: rmudgett
+ Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged
+ revisions 375964 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-12-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.11.0 Released.
+
+2012-12-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.11.0-rc3 Released.
+
+ * chan_local: Fix local_pvt ref leak in local_devicestate().
+
+ Regression introduced by ASTERISK-20390 fix.
+
+ (closes issue ASTERISK-20769)
+ Reported by: rmudgett
+
+2012-12-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.11.0-rc2 Released.
+
+ * Fix a SIP request memory leak with TLS connections.
+
+ During the TLS re-work in chan_sip some TLS specific code was moved
+ into a separate function. This function operates on a copy of the
+ incoming SIP request. This copy was never deinitialized causing a
+ memory leak for each request processed.
+
+ This function is now given a SIP request structure which it can use
+ to copy the incoming request into. This reduces the amount of memory
+ allocations done since the internal allocated components are reused
+ between packets and also ensures the SIP request structure is
+ deinitialized when the TLS connection is torn down.
+
+ (closes issue ASTERISK-20763)
+ Reported by: deti
+
+2012-11-06 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.11.0-rc1 Released.
+
+2012-11-05 23:00 +0000 [r375894] Matthew Jordan <mjordan at digium.com>
+
+ * main/timing.c, main/channel.c, /, res/res_timing_pthread.c,
+ res/res_timing_dahdi.c, res/res_timing_timerfd.c,
+ bridges/bridge_softmix.c, funcs/func_jitterbuffer.c,
+ include/asterisk/timing.h, res/res_musiconhold.c,
+ channels/chan_iax2.c, res/res_fax_spandsp.c,
[... 28219 lines stripped ...]
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