[asterisk-commits] bebuild: tag 11.6.0-rc1 r399452 - /tags/11.6.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 19 09:19:26 CDT 2013
Author: bebuild
Date: Thu Sep 19 09:19:24 2013
New Revision: 399452
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=399452
Log:
Importing files for 11.6.0-rc1 release.
Added:
tags/11.6.0-rc1/.lastclean (with props)
tags/11.6.0-rc1/.version (with props)
tags/11.6.0-rc1/ChangeLog (with props)
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+2013-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.6.0-rc1 Released.
+
+2013-09-18 23:36 +0000 [r399442] Richard Mudgett <rmudgett at digium.com>
+
+ * main/udptl.c: UDPTL: Backport some fixes from v12 that should be
+ in v11. Backported the following as applied to udptl.c: *
+ -r398020 Fixup udpdl defaults if config file not present. *
+ -r398533 Fixup improper use of ao2_global_obj_replace().
+
+2013-09-18 19:55 +0000 [r399403] Kinsey Moore <kmoore at digium.com>
+
+ * main/abstract_jb.c, /: Fix jitter buffer log file creation This
+ adjusts '/'-to-'#' replacement to replace all instances of '/'
+ instead of just the first to ensure that the jitter buffer log
+ file gets the correct name as per Richard Kenner's suggestion.
+ (closes issue ASTERISK-21036) Reported by: Richard Kenner
+ ........ Merged revisions 399402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-18 17:22 +0000 [r399353-399373] Matthew Jordan <mjordan at digium.com>
+
+ * /, build_tools/prep_tarball: Update prep_tarball with new
+ documentation files on the Asterisk wiki This will now pull both
+ a command reference for the version being prepared, as well as an
+ Admin Guide that applies to all versions of Asterisk. (issue
+ ASTERISK-22439) Reported by: Olle Johansson ........ Merged
+ revisions 399351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when a
+ timing module isn't loaded If bridge_softmix fails to be created
+ because no timing source is present in Asterisk, this will
+ currently fail gracefully but with (most likely) a generic error
+ message by whatever module tried to create the softmix bridge.
+ This patch adds a more explicit warning so you can actually
+ diagnose and fix the problem. Review:
+ https://reviewboard.asterisk.org/r/2857/
+
+2013-09-18 01:34 +0000 [r399305] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, main/features.c: Fix Segfault When Syntax Of A Line Under
+ [applicationmap] Is Invalid When processing the lines under the
+ [applicationmap] context in features.conf, a segfault occurs from
+ attempting to process a line with an invalid syntax (basically
+ missing most of the arguments). Example: [applicationmap]
+ automon=*6 * This patch moves the checking for empty arguments to
+ before they are accessed. * Also, checked the "todo" comment and
+ removed it. Some applications do not require arguments. (closes
+ issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
+ Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2803 ........ Merged revisions
+ 399304 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-17 18:32 +0000 [r399222-399267] Kevin Harwell <kharwell at digium.com>
+
+ * main/asterisk.c, main/logger.c: Remote console: more output
+ discrepancies The remote console continued to have issues with
+ its output. In this case CLI command output would either not show
+ up (if verbose level = 0) or would contain verbose prefixes (if
+ verbose level > 0) once log messages were sent to the remote
+ console. The fix now now adds verbose prefix data to all new
+ lines contained in a verbose log string. (closes issue
+ ASTERISK-22450) Reported by: David Brillert (closes issue
+ AST-1193) Reported by: Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2825/
+
+ * apps/confbridge/conf_state_multi_marked.c: Confbridge: empty
+ conference not being torn down Confbridge would not properly tear
+ down an empty conference bridge when all users were kicked via
+ end_marked=yes and at least one user was also set to wait_marked.
+ This occurred because while end_marked users were being kicked
+ and at least one was also set to wait_marked then the leave
+ wait_marked handler would be called on that user, but there would
+ be no waiting user (still considered active). The waiting users
+ would decrement and now be negative. The conference would remain,
+ but be put into an inactive state. The solution was to move from
+ the active list to the wait list, those users with wait_marked
+ set right before kicking. This allows both the active and wait
+ users to decrement correctly and the confbridge to tear down
+ properly. A crashed also occurred when trying to list the
+ specific conference from the CLI. This happened because the
+ conference specified was invalid. Since the conference properly
+ tears down now there is no way to reference it thus alleviating
+ the crash as well. (closes issue ASTERISK-21859) Reported by:
+ Chris Gentle Review: https://reviewboard.asterisk.org/r/2848/
+
+2013-09-16 16:42 +0000 [r399159] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix saving the wrong expiry
+ time in astdb. When a new IAX2 client registers, the astdb
+ database is updated with the value of minregexpire defined in
+ iax.conf instead of using the expiry time that is provided by the
+ client. The provided expiry time of the client is updated after
+ inserting the astdb entry. As a consequence, restarting or
+ reloading asterisk creates clients whose registration may expire
+ before they reregister. The clients are therefore unavailable
+ after minregexpire seconds until they reregister. * Move updating
+ of the expiry time to before inserting into the astdb. (closes
+ issue ASTERISK-22504) Reported by: Stefan Wachtler Patches:
+ chan_iax2.c.patch (license #6533) patch uploaded by Stefan
+ Wachtler ........ Merged revisions 399158 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-13 20:49 +0000 [r399099] David M. Lee <dlee at digium.com>
+
+ * main/astobj2.c, /: Don't write to /tmp/refs when REF_DEBUG is not
+ defined. If MALLOC_DEBUG is enabled, then the debug destructor
+ for the container is used, which would erroneously write to
+ /tmp/refs. This patch only uses the debug destructor if ref_debug
+ is used. (closes issue ASTERISK-22536) ........ Merged revisions
+ 399098 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-13 13:48 +0000 [r399034] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_meetme.c: Fix several crashes in MeetMeAdmin This
+ change ensures that MeetMeAdmin commands requiring a user
+ actually get a user and fixes another issue where an extra
+ dereference could occur for a last-entered user being ejected if
+ a user identifier was also provided. (closes issue
+ ASTERISK-21907) Reported by: Alex Epshteyn Review:
+ https://reviewboard.asterisk.org/r/2844/ ........ Merged
+ revisions 399033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-12 20:19 +0000 [r398986] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Revert r398835 due to failing tests involving originate (issue
+ ASTERISK-22424) Reported by: Jonathan Rose ........ Merged
+ revisions 398977 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-12 00:02 +0000 [r398881-398885] Rusty Newton <rnewton at digium.com>
+
+ * /, apps/app_queue.c: 'queue add member' help text correction You
+ are adding dial strings to the queue, not channels. An aribitrary
+ string could be used, but you are typically referencing a
+ channel. Correcting the command help text. (issue ASTERISK-22263)
+ (closes issue ASTERISK-22263) Reported By: Rusty Newton ........
+ Merged revisions 398884 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/chan_dahdi.conf.sample, /: Documentation fix -
+ waitfordialtone is not boolean, it's time in milliseconds
+ Changing text in chan_dahdi.conf sample to be accurate. (issue
+ ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
+ Malcolm Davenport ........ Merged revisions 398880 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-11 19:46 +0000 [r398836] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/include/sip.h, /, channels/chan_sip.c: chan_sip:
+ Reject calls without prior SDP on 200 OK If we receive a 200 OK
+ without SDP, we will now check to see if the remote address has
+ been established for that channel's RTP session and if the to tag
+ for that channel has changed from the most recent to tag in a
+ response less than 200. If either a change has been made since
+ the last to-tag was received or the remote address is unset, then
+ we will drop the call. (closes issue ASTERISK-22424) Reported by:
+ Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/diff/#index_header
+ ........ Merged revisions 398835 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-11 18:01 +0000 [r398820] Russell Bryant <russell at russellbryant.com>
+
+ * configs/confbridge.conf.sample: Fix typo in
+ confbridge.conf.sample The denoise filter requires func_speex,
+ not codec_speex. Fix this in the description of the denoise=yes
+ option in confbridge.conf.
+
+2013-09-10 17:56 +0000 [r398758] Richard Mudgett <rmudgett at digium.com>
+
+ * main/event.c, res/res_musiconhold.c, main/indications.c,
+ main/asterisk.c, main/xmldoc.c, main/cli.c, /,
+ funcs/func_dialgroup.c, main/heap.c: Fix incorrect usages of
+ ast_realloc(). There are several locations in the code base where
+ this is done: buf = ast_realloc(buf, new_size); This is going to
+ leak the original buf contents if the realloc fails. Review:
+ https://reviewboard.asterisk.org/r/2832/ ........ Merged
+ revisions 398757 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-10 17:48 +0000 [r398749-398753] David M. Lee <dlee at digium.com>
+
+ * utils/check_expr.c, /: Fixed utils directory breakage from
+ r398748, this time with extra hate. ........ Merged revisions
+ 398752 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c, /: Fixed
+ utils directory breakage from r398648 ........ Merged revisions
+ 398748 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-09 23:21 +0000 [r398721] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
+ completely different from the freed magic number. Race conditions
+ between freeing a nul terminated string and ast_strdup()'ing it
+ are more likely to be detected if the fence and freed magic
+ numbers are completely different. ........ Merged revisions
+ 398703 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-09 20:02 +0000 [r398649] David M. Lee <dlee at digium.com>
+
+ * main/lock.c, /, main/utils.c, include/asterisk/lock.h: Fix
+ DEBUG_THREADS when lock is acquired in __constructor__ This patch
+ fixes some long-standing bugs in debug threads that were
+ exacerbated with recent Optional API work in Asterisk 12. With
+ debug threads enabled, on some systems, there's a lock ordering
+ problem between our mutex and glibc's mutex protecting its module
+ list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
+ thread, the module list will be locked before acquiring our
+ mutex. In another thread, our mutex will be locked before locking
+ the module list (which happens in the depths of calling
+ backtrace()). This patch fixes this issue by moving backtrace()
+ calls outside of critical sections that have the mutex acquired.
+ The bigger change was to reentrancy tracking for
+ ast_cond_{timed,}wait, which wrongly assumed that waiting on the
+ mutex was equivalent to a single unlock (it actually suspends all
+ recursive locks on the mutex). (closes issue ASTERISK-22455)
+ Review: https://reviewboard.asterisk.org/r/2824/ ........ Merged
+ revisions 398648 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-07 00:59 +0000 [r398510-398618] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_xmpp.c: Prevent XMPP timeout on blank responses Sometimes
+ the Google Voice servers have a bad habit of sending out 1 byte
+ replies to the xmpp resource. When a blank 1 byte reply is
+ received from the socket the buffer attempts to wait (endlessly)
+ for the rest of the reply from google which effectively blocks
+ the socket and google voice calls will no longer come into the
+ server. This patch allows the xmpp module to correctly detect
+ empty packets and send out ping replies to google. It also sets a
+ socket timeout on the default socket which prevents the xmpp
+ socket from closing and preventing future google voice calls from
+ coming into the server. Furthermore instead of sending an empty
+ reply back to google we send a proper xmpp ping reply back. This
+ also adds several more socket messages. (closes issue
+ ASTERISK-22347) Reported by: Andrew Nagy Review:
+ https://reviewboard.asterisk.org/r/2771 Patches: xmpp_fix_1.diff
+ uploaded by Andrew Nagy (License #6524)
+
+ * /, res/res_xmpp.c, res/res_jabber.c: Commit the remainder of
+ r398523 This is a missing part of the commit in revision 398523
+ that corrects the name of a variable. (issue ASTERISK-22435)
+ ........ Merged revisions 398576 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, res/res_xmpp.c, res/res_jabber.c: Fix Jabber/XMPP distributed
+ MWI The mailbox and context are swapped on the receiving end for
+ all users of Jabber and XMPP distributed MWI in Asterisk 1.8 and
+ all more recent versions. This swaps those values to be correct
+ when publishing to the internal event system from Jabber/XMPP
+ distributed MWI state. (closes issue ASTERISK-22435) Reported by:
+ abelbeck Tested by: Michael Keuter Patches:
+ asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
+ abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
+ uploaded by abelbeck ........ Merged revisions 398523 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_h323.c: Fix chan_h323 compilation This fixes the
+ things in chan_h323 that were missed or ignored in the great
+ channel opaquification and gets chan_h323 back into a compiling
+ state. (closes issue ASTERISK-22365) Reported by: Dmitry Melekhov
+ Patches: chan_h323.patch uploaded by Dmitry Melekhov
+
+2013-09-05 19:13 +0000 [r398302-398457] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, /: chan_iax2: Reduce indentation in
+ __attempt_transmit(). * Reduce indentation in
+ __attempt_transmit(). * Don't update the static last error time
+ variable every time in __schedule_action() and socket_read().
+ ........ Merged revisions 398456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix stray reference to worker
+ thread idle_list. * Fix stray reference to idle_list in
+ cleanup_thread_list(). This may be the reason for the note in
+ iax2_process_thread() about threads not being removed from the
+ task lists. * Move cleanup_thread_list(&idle_list) to after the
+ other lists are cleaned up. ........ Merged revisions 398416 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_iax2.c, /: chan_iax2: Fix bridgecallno deadlock
+ avoidance. * Fix bridgecallno deadlock avoidance. When doing
+ deadlock avoidance, you need to retest the status of values for
+ each loop to see if you still need the lock for bridgecallno. *
+ As a safety check, after acquiring the bridgecallno lock you
+ should check if iaxs[bridgecallno] is NULL just like the current
+ callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
+ to after processing any deferred frames to ensure that the
+ iostate is IDLE when it is placed back into the idle list.
+ defer_full_frame() tries to ensure iax2_process_thread() wakes up
+ to process the frame. ........ Merged revisions 398379 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/iax2-parser.c: chan_iax2: Add missing control frame
+ names to debug frame decode output. (Part 2)
+
+ * channels/iax2-parser.c, /: chan_iax2: Add missing control frame
+ names to debug frame decode output. ........ Merged revisions
+ 398301 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-04 21:33 +0000 [r398281-398285] Jonathan Rose <jrose at digium.com>
+
+ * tests/test_voicemail_api.c: unit tests: test_voicemail_api leaks
+ stringfields from snapshots (closes issue ASTERISK-22414)
+ Reported by: Corey Farrell Patches:
+ test_voicemail_api-leaks-11.patch uploaded by coreyfarrell
+ (license 5909)
+
+ * apps/app_voicemail.c: app_voicemail: Fix leaking config objects
+ when msg_id doesn't match (issues ASTERISK-22414) Reported by:
+ Corey Farrell Patch: test_voicemail_api-leaks-11.patch uploaded
+ by coreyfarrell (license 5909)
+
+2013-09-04 15:57 +0000 [r398236] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_misdn.c, /: chan_misdn: Fix misdn debug output
+ printed with arbitrary verbose levels. Fix the misdn debug output
+ to remote consoles. chan_misdn uses ast_console_puts() which
+ doesn't know about verbose levels. Better to use ast_verbose()
+ instead. Without this patch the misdn debug messages are appended
+ to the verbose level which ever was set by the message sent to
+ the console before, i.e. any undefined level. (closes issue
+ AST-1218) Reported by: Guenther Kelleter Patches: misdnlog.patch
+ (license #6372) patch uploaded by Guenther Kelleter ........
+ Merged revisions 398235 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-03 19:45 +0000 [r398214] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooh245.c: Fix remote tcs sequence handling on
+ empty tcs received
+
+2013-09-02 07:28 +0000 [r398168] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, cel/cel_custom.c: Be a little more verbose when loading
+ cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
+ ........ Merged revisions 398167 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 19:16 +0000 [r398022-398103] Kevin Harwell <kharwell at digium.com>
+
+ * main/indications.c, main/config.c, res/res_security_log.c, /,
+ channels/chan_sip.c, main/translate.c, main/named_acl.c: Fix
+ various memory leaks main/config.c - cleanup cache fie includes
+ res/res_security_log.c - unregister logger level
+ channesl/chan_sip.c - cleanup io context and notify_types
+ main/translator.c - cleanup at shutdown main/named_acl.c -
+ cleanup cli commands main/indications.c -
+ ast_get_indication_tone() unref default_tone_zone if used (closes
+ issues ASTERISK-22378) Reported by: Corey Farrell Patches:
+ config_shutdown.patch uploaded by coreyfarrell (license 5909)
+ res_security_log.patch uploaded by coreyfarrell (license 5909)
+ chan_sip-11.patch uploaded by coreyfarrell (license 5909)
+ indications_refleak.patch uploaded by coreyfarrell (license 5909)
+ named_acl-cli_unreg-11.patch uploaded by coreyfarrell (license
+ 5909) translate_shutdown.patch uploaded by coreyfarrell (license
+ 5909) ........ Merged revisions 398102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_agi.c, main/manager.c, /: Memory leak fix
+ ast_xmldoc_printable returns an allocated block that must be
+ freed by the caller. Fixed manager.c and res_agi.c to stop
+ leaking these results. (closes issue ASTERISK-22395) Reported by:
+ Corey Farrell Patches: manager-leaks-11.patch uploaded by
+ coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+ by coreyfarrell (license 5909) ........ Merged revisions 398060
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix memory leak Fixed a features.c test that
+ leaked a reference to a parked call. This caused chancount to
+ never reach 0, so graceful shutdown stops. Also added an
+ unregister test. (closes issue ASTERISK-22413) Reported by: Corey
+ Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
+ coreyfarrell (license 5909) ........ Merged revisions 398021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 16:57 +0000 [r398019] Richard Mudgett <rmudgett at digium.com>
+
+ * tests/test_substitution.c, /: test_substituition: Fix failed test
+ reporting to actually report failure. You cannot put the "Testing
+ <blah> pass/fail" on a single line before actually performing the
+ test. Now any additional failure information is logged before the
+ test pass/fail announcement. * Added an additional CDR(answer,u)
+ test. ........ Merged revisions 398018 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-30 16:20 +0000 [r397948-398011] Kevin Harwell <kharwell at digium.com>
+
+ * /, apps/app_mixmonitor.c: Fix memory leaks (closes issue
+ ASTERISK-22368) Reported by: Corey Farrell Patches:
+ issueA22368_mixmonitor_free_filename.patch uploaded by wdoekes
+ (license 5674) ........ Merged revisions 398004 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+ * main/asterisk.c: Check return value on fwrite
+
+ * channels/chan_misdn.c, apps/app_dumpchan.c, main/features.c,
+ main/logger.c, apps/app_verbose.c, main/asterisk.c: Verbose
+ logging discrepancies Refactored cases where a combination of
+ ast_verbose/options_verbose were present. Also in general tried
+ to eliminate, in as many places as possible, where the
+ options_verbose global variable was being used. Refactored the
+ way local and remote consoles handle verbose message logging in
+ an attempt to solve the various discrepancies that sometimes
+ would show between the two. (closes issue AST-1193) Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/2798/
+
+2013-08-27 18:03 +0000 [r397758] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
+ SDP If the SIP channel driver processes an invalid SDP that
+ defines media descriptions before connection information, it may
+ attempt to reference the socket address information even though
+ that information has not yet been set. This will cause a crash.
+ This patch adds checks when handling the various media
+ descriptions that ensures the media descriptions are handled only
+ if we have connection information suitable for that media. Thanks
+ to Walter Doekes, OSSO B.V., for reporting, testing, and
+ providing the solution to this problem. (closes issue
+ ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
+ issueA22007_sdp_without_c_death.patch uploaded by wdoekes
+ (License 5674) ........ Merged revisions 397756 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397757 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-08-27 16:40 +0000 [r397744] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c, channels/chan_motif.c, channels/chan_iax2.c,
+ channels/sig_pri.c, channels/sig_ss7.c, channels/chan_dahdi.c,
+ channels/sig_analog.c: Fix uninitialized value in struct
+ ast_control_pvt_cause_code usage.
+
+2013-08-27 15:55 +0000 [r397712] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK
+ on dialog that has no channel A remote exploitable crash
+ vulnerability exists in the SIP channel driver if an ACK with SDP
+ is received after the channel has been terminated. The handling
+ code incorrectly assumed that the channel would always be
+ present. This patch adds a check such that the SDP will only be
+ parsed and applied if Asterisk has a channel present that is
+ associated with the dialog. Note that the patch being applied was
+ modified only slightly from the patch provided by Walter Doekes
+ of OSSO B.V. (closes issue ASTERISK-21064) Reported by: Colin
+ Cuthbertson Tested by: wdoekes, Colin Cutherbertson patches:
+ issueA21064_fix.patch uploaded by wdoekes (License 5674) ........
+ Merged revisions 397710 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+ revisions 397711 from
+ http://svn.asterisk.org/svn/asterisk/branches/10
+
+2013-08-23 21:57 +0000 [r397604] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c, UPGRADE.txt, res/Makefile: Make libuuid
+ an optional dependency for res_rtp_asterisk instead of a
+ requirement. Review: https://reviewboard.asterisk.org/r/2777/
+
+2013-08-23 16:07 +0000 [r397528] Richard Mudgett <rmudgett at digium.com>
+
+ * main/utils.c, include/asterisk/lock.h, main/astmm.c,
+ channels/sig_pri.c, main/astobj2.c, include/asterisk/logger.h,
+ main/lock.c, include/asterisk/utils.h, include/asterisk/astmm.h,
+ /, main/logger.c: Fix memory corruption when trying to get "core
+ show locks". Review https://reviewboard.asterisk.org/r/2580/
+ tried to fix the mismatch in memory pools but had a math error
+ determining the buffer size and didn't address other similar
+ memory pool mismatches. * Effectively reverted the previous patch
+ to go in the same direction as trunk for the returned memory pool
+ of ast_bt_get_symbols(). * Fixed memory leak in
+ ast_bt_get_symbols() when BETTER_BACKTRACES is defined. * Fixed
+ some formatting in ast_bt_get_symbols(). * Fixed sig_pri.c
+ freeing memory allocated by libpri when MALLOC_DEBUG is enabled.
+ * Fixed __dump_backtrace() freeing memory from
+ ast_bt_get_symbols() when MALLOC_DEBUG is enabled. * Moved
+ __dump_backtrace() because of compile issues with the utils
+ directory. (closes issue ASTERISK-22221) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/2778/ ........ Merged
+ revisions 397525 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-22 08:22 +0000 [r397378] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * default.exports, /, main/asterisk.exports.in: Add _IO_stdin_used
+ in version-script to fix SIGBUSes on Sparc. The
+ --version-script,asterisk.exports linker flag (and the module
+ exports) didn't provide _IO_stdin_used in the list of exported
+ symbols. That causes some kind of libc compatibility mode to kick
+ in, where stdio file structures (stdout/stderr) land somewhere
+ else. In the case of the Sparc, they landed on misaligned memory.
+ This became apparent first after r376428 (Reorder startup
+ sequence) when a lot of ast_log's were replaced with fprintf's.
+ Writing to stderr triggered a SIGBUS. (Compared to x86 and amd64
+ architectures, the Sparc is very picky about memory alignment.)
+ (issue ASTERISK-21763) (issue ASTERISK-21665) Reported by: Jeremy
+ Kister Review: https://reviewboard.asterisk.org/r/2760/ ........
+ Merged revisions 397377 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 23:02 +0000 [r397365] Jonathan Rose <jrose at digium.com>
+
+ * main/udptl.c: UDPTL: Fix a regression where UDPTL won't load
+ default settings If the file udptl.conf is unavailable at
+ startup, UDPTL will fail to initialize and while it makes some
+ noise, it isn't immediately obvious why consumers start to fail
+ when using it. This patch makes UDPTL load as though an empty
+ config was provided when udptl is unavailable at startup. (closes
+ issue ASTERISK-22349) Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2773/
+
+2013-08-21 17:07 +0000 [r397309] David M. Lee <dlee at digium.com>
+
+ * /, main/http.c: Complete http_shutdown. This patch frees up some
+ resources allocated in http.c. * tcp listeners stopped * tls
+ settings freed * uri redirects freed * unregister internal http.c
+ uri's (closes issue ASTERISK-22237) Reported by: Corey Farrell
+ Patches: http.patch uploaded by Corey Farrell (license 5909)
+ ........ Merged revisions 397308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 15:12 +0000 [r397257] Matthew Jordan <mjordan at digium.com>
+
+ * /, include/asterisk/frame.h: Set 14400 as the default max bit
+ rate if T38MaxBitRate is not specified If an endpoint fails to
+ include the T38MaxBitRate attribute during negotiation, Asterisk
+ will negotiate a bit rate of 2400 instead of the ITU recommended
+ bit rate of 14400. This patch fixes this by making
+ AST_T38_RATE_14400 the 'default' value of the enum by assigning
+ it a value of 0, such that if an endpoint fails to include the
+ attribute, the default will be 14400. Note that Walter Doekes
+ included the nice comment in frame.h about why we are
+ purposefully assigning AST_T38_RATE_14400 a value of 0. (closes
+ issue ASTERISK-22275) Reported by: Andreas Steinmetz patches:
+ fax-fix.patch uploaded by anstein (License 6523) ........ Merged
+ revisions 397256 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-21 14:36 +0000 [r397254] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Prevent a crash on outbound SIP MESSAGE
+ requests. If a From header on an outbound out-of-call SIP MESSAGE
+ were malformed, the result could crash Asterisk. In addition, if
+ a From header on an incoming out-of-call SIP MESSAGE request were
+ malformed, the message was happily accepted rather than being
+ rejected up front. The incoming message path would not result in
+ a crash, but the behavior was bad nonetheless. (closes issue
+ ASTERISK-22185) reported by Zhang Lei
+
+2013-08-21 02:11 +0000 [r397205] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, channels/chan_sip.c: Fix Not Storing Current Incoming Recv
+ Address In 1.8, r384779 introduced a regression by retrieving an
+ old dialog and keeping the old recv address since recv was
+ already set. This has caused a problem when a proxy is involved
+ since responses to incoming requests from the proxy server, after
+ an outbound call is established, are never sent to the correct
+ recv address. In 11, r382322 introduced this regression. The fix
+ is to revert that change and always store the recv address on
+ incoming requests. Thank you Walter Doekes for helping to point
+ out this error and Mark Michelson for your input/review of the
+ fix. (closes issue ASTERISK-22071) Reported by: Alex Zarubin
+ Tested by: Alex Zarubin, Karsten Wemheuer Patches:
+ asterisk-22071-store-recvd-address.diff by Michael L. Young
+ (license 5026) ........ Merged revisions 397204 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 17:41 +0000 [r397133-397157] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Remove REF_DEBUG definition. ........
+ Merged revisions 397156 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, channels/sip/dialplan_functions.c: Fix
+ refcounting of sip_pvt in test_sip_rtpqos test and unlink it from
+ the list of pvts. (closes issue ASTERISK-22248) reported by Corey
+ Farrell patches: test_sip_rtpqos.patch uploaded by Corey Farrell
+ (license #5909) ........ Merged revisions 397112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 15:27 +0000 [r397034-397107] Kinsey Moore <kmoore at digium.com>
+
+ * /, main/threadstorage.c, main/astfd.c: Unregister CLI commands on
+ exit This patch ensures that CLI commands enabled by
+ DEBUG_FD_LEAKS and DEBUG_THREADLOCALS are cleaned up properly on
+ exit. (closes issue ASTERISK-22238) Reported by: Corey Farrell
+ Tested by: Corey Farrell Patches: debug_cli_unregister.patch
+ uploaded by Corey Farrell ........ Merged revisions 397106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/xmldoc.c, /: Fix xmldoc memory leak This fixes a
+ single-attribute memory leak that was occurring when the
+ "required" attribute was not true. (closes issue ASTERISK-22249)
+ Reported by: Corey Farrell Tested by: Corey Farrell Patches:
+ xmldoc-free_attr_required.patch uploaded by Corey Farrell
+ ........ Merged revisions 397064 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/cel.c, /: Protect CEL from an invalid config on reload This
+ patch fixes CEL to properly handle an invalid config on reload.
+ (closes issue ASTERISK-22259) Reported by: Corey Farrell Tested
+ by: Corey Farrell Patches: cel-config.patch uploaded by Corey
+ Farrell ........ Merged revisions 397033 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 11:47 +0000 [r396995] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * configs/h323.conf.sample, /, configs/sip.conf.sample: Add
+ "autoframing" option to sip.conf.sample and h323.conf.sample. The
+ autoframing option was added to chan_sip.c in r43243 (mogorman,
+ 2006-09-19 01:32:57), but never made its way into the sample
+ configs. Review: https://reviewboard.asterisk.org/r/2768/
+ ........ Merged revisions 396994 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-20 01:18 +0000 [r396944-396961] Matthew Jordan <mjordan at digium.com>
+
+ * main/data.c, /: Fix invalid access to disposed memory in
+ main/data unit test It is not safe to iterate over a macro'd list
+ of ao2 objects, deref them such that the item's destructor is
+ called, and leave them in the list. The list macro to iterate
+ over items requires the item to be a valid allocated object in
+ order to proceed to the next item; with MALLOC_DEBUG on the
+ corruption of the linked list is caught in the crash. This patch
+ fixes the invalid access to free'd memory by removing the ao2
+ item from the list before de-refing it. Note that this is a
+ backport of r396915 from Asterisk trunk. ........ Merged
+ revisions 396958 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c: Let Queue wrap up time influence member
+ availability Queue members who happen to be in multiple queues at
+ the same time may not have any wrap up time. This problem
+ occurred due to a code change in Asterisk 11.3.0 that unified
+ device state tracking of Queue members in multiple Queues (which
+ fixed some other problems, but unfortunately caused this one).
+ This patch fixes the behavior by having the is_member_available
+ function check the queue's wrap up time and the time of the
+ member's last call, such that for a particular queue, the member
+ won't be considered available if their last call is within the
+ wrap up time. (closes issue ASTERISK-22189) Reported by: Tony
+ Lewis Tested by: Tony Lewis
+
+ * apps/app_meetme.c: Resolve conflicts between
+ CONFFLAG_DONT_DENOISE and CONFFLAG_INTROUSER_VMREC When r382230
+ added an option to not denoise the MeetMe conference (if a user
+ had a channel whose format's sample rate changed frequently, for
+ example), the value added was the maximum allowed value for the
+ constants that define the options for MeetMe in 1.8. Not so in 11
+ - unfortunately, the option CONFFLAG_DONT_DENOISE conflicts with
+ CONFFLAG_INTROUESR_VMREC. This patch fixes that, and also tweaks
+ one of the way in which the constants was declared for
+ consistency. Thanks to Tony Mountifield for pointing out the
+ problem and solution. (closes issue ASTERISK-22269) Reported by:
+ Tony Mountifield
+
+2013-08-16 22:45 +0000 [r396884] John Bigelow <jbigelow at digium.com>
+
+ * main/features.c: Add test suite events to indicate when a feature
+ is detected or not These are needed by the bridge test suite
+ tests for them to be able to run against Asterisk 11. Review:
+ https://reviewboard.asterisk.org/r/2751/
+
+2013-08-15 16:29 +0000 [r396746] Kinsey Moore <kmoore at digium.com>
+
+ * main/asterisk.c, main/cli.c, /: Remove leading spaces from the
+ CLI command before parsing If you've mistakenly put a space
+ before typing in a command, the leading space will be included as
+ part of the command, and the command parser will not find the
+ corresponding command. This patch rectifies that situation by
+ stripping the leading spaces on commands. Review:
+ https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman
+ Lesher ........ Merged revisions 396745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-14 19:06 +0000 [r396620-396657] Joshua Colp <jcolp at digium.com>
+
+ * tests/test_hashtab_thrash.c, /: Tweak comment for why usleep is
+ used. ........ Merged revisions 396656 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, tests/test_hashtab_thrash.c: Tweak test_hashtab_thrash test to
+ allow the critical threads to execute. Depending on certain
+ conditions it was possible for the hashtab counting thread to
+ starve other threads, preventing them from executing in the
+ expected fashion. This change adds a sleep to allow the others to
+ do what they need to do. While this doesn't thrash the hashtab as
+ much as previously, it at least works. (closes issue
+ ASTERISK-22276) Reported by: Matt Jordan ........ Merged
+ revisions 396619 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-08-13 18:45 +0000 [r396580-396583] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, channels/chan_sip.c: chan_sip: Convert 'just did sched_add
+ waitid...' from warning to debug message. Patches:
+ reviewboard-2377.patch uploaded by Paul Belanger Review:
+ https://reviewboard.asterisk.org/r/2377/ ........ Merged
+ revisions 396582 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
+ rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
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