[svn-commits] bebuild: tag 1.8.24.0-rc1 r399446 - /tags/1.8.24.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Sep 19 09:03:58 CDT 2013
Author: bebuild
Date: Thu Sep 19 09:03:56 2013
New Revision: 399446
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=399446
Log:
Importing files for 1.8.24.0-rc1 release.
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tags/1.8.24.0-rc1/.version (with props)
tags/1.8.24.0-rc1/ChangeLog (with props)
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+2013-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.24.0-rc1 Released.
+
+2013-09-18 19:54 +0000 [r399402] 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
+
+2013-09-18 17:15 +0000 [r399351] 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
+
+2013-09-18 01:32 +0000 [r399304] 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
+
+2013-09-16 16:37 +0000 [r399158] 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
+
+2013-09-13 20:47 +0000 [r399098] 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)
+
+2013-09-13 13:31 +0000 [r399033] 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/
+
+2013-09-12 20:09 +0000 [r398937-398977] 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
+
+ * res/res_musiconhold.c: res_musiconhold: Fix reference leaks
+ caused when reloading with REF_DEBUG set Due to a faulty function
+ for debugging reference decrementing, it was possible to reduce
+ the refcount on the wrong object if two moh classes of the same
+ name were in the moh class container. (closes issue
+ ASTERISK-22252) Reported by: Walter Doekes Patches:
+ 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
+ 6182)
+
+2013-09-12 00:00 +0000 [r398880-398884] 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
+
+ * 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
+
+2013-09-11 19:39 +0000 [r398835] 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
+
+2013-09-10 17:53 +0000 [r398757] Richard Mudgett <rmudgett at digium.com>
+
+ * main/xmldoc.c, main/cli.c, funcs/func_dialgroup.c, main/heap.c,
+ main/event.c, res/res_musiconhold.c, main/indications.c,
+ main/asterisk.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/
+
+2013-09-10 17:47 +0000 [r398748-398752] David M. Lee <dlee at digium.com>
+
+ * utils/check_expr.c: Fixed utils directory breakage from r398748,
+ this time with extra hate.
+
+ * utils/check_expr.c, utils/ael_main.c, utils/conf2ael.c: Fixed
+ utils directory breakage from r398648
+
+2013-09-09 23:15 +0000 [r398703] 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.
+
+2013-09-09 19:56 +0000 [r398648] David M. Lee <dlee at digium.com>
+
+ * main/utils.c, include/asterisk/lock.h, main/lock.c: 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/
+
+2013-09-06 20:56 +0000 [r398523-398576] Kinsey Moore <kmoore at digium.com>
+
+ * 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)
+
+ * 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
+
+2013-09-05 19:00 +0000 [r398235-398456] 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().
+
+ * 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.
+
+ * 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.
+
+ * 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.
+
+ * 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
+
+2013-09-02 07:24 +0000 [r398167] 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/
+
+2013-08-30 18:55 +0000 [r398021-398102] Kevin Harwell <kharwell at digium.com>
+
+ * channels/chan_sip.c, main/config.c, res/res_security_log.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 (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-1.8.patch uploaded by coreyfarrell (license 5909)
+
+ * main/manager.c, res/res_agi.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-1.8.patch uploaded by
+ coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+ by coreyfarrell (license 5909)
+
+ * 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)
+
+2013-08-30 16:46 +0000 [r398018] 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.
+
+2013-08-27 17:55 +0000 [r397710-397756] 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)
+
+ * 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)
+
+2013-08-23 15:34 +0000 [r397525] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/astmm.h, main/logger.c, 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: 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/
+
+2013-08-22 08:19 +0000 [r397377] 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/
+
+2013-08-21 17:00 +0000 [r397308] 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)
+
+2013-08-21 14:56 +0000 [r397256] 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)
+
+2013-08-21 02:09 +0000 [r397204] 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)
+
+2013-08-20 17:40 +0000 [r397112-397156] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Remove REF_DEBUG definition.
+
+ * channels/sip/dialplan_functions.c, channels/chan_sip.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)
+
+2013-08-20 15:26 +0000 [r397033-397106] 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
+
+ * 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
+
+ * 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
+
+2013-08-20 11:46 +0000 [r396994] 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/
+
+2013-08-20 01:17 +0000 [r396958] 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.
+
+2013-08-15 16:21 +0000 [r396745] Kinsey Moore <kmoore at digium.com>
+
+ * main/cli.c, main/asterisk.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
+
+2013-08-14 19:05 +0000 [r396619-396656] Joshua Colp <jcolp at digium.com>
+
+ * tests/test_hashtab_thrash.c: Tweak comment for why usleep is
+ used.
+
+ * 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
+
+2013-08-13 18:44 +0000 [r396579-396582] 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/
+
+ * channels/chan_sip.c: chan_sip: Fix IP-addr in warning when
+ rejecting a contact ACL. Patches: reviewboard-2155.patch uploaded
+ by Paul Belanger Review: https://reviewboard.asterisk.org/r/2155/
+
+2013-08-08 20:14 +0000 [r396427] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/logger.c, main/utils.c, main/astobj2.c,
+ include/asterisk/logger.h: Consistent memory allocation by
+ ast_bt_get_symbols. Always use ast_alloc/ast_free. This is
+ handled differently in trunk (r391012). Review:
+ https://reviewboard.asterisk.org/r/2580/
+
+2013-08-06 08:14 +0000 [r396279] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * pbx/pbx_dundi.c, utils/extconf.c, apps/app_stack.c,
+ apps/app_playback.c, funcs/func_global.c, main/cdr.c,
+ pbx/pbx_loopback.c, main/pbx.c, funcs/func_strings.c: Check
+ result of ast_var_assign() calls for memory allocation failure.
+ We try to keep the system running even when all available memory
+ is spent. Review: https://reviewboard.asterisk.org/r/2734/
+
+2013-08-05 20:17 +0000 [r396196-396240] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Fix Registration Failure When A Peer And TLS
+ Are Used If a peer is used in a register line and TLS is defined
+ as the transport, the registration fails since the transport on
+ the dialog is never set properly resulting in UDP being used
+ instead of TLS. This patch sets the dialog's transport based on
+ the transport that was defined in the register line. If the
+ register line does not specify a transport, the parsing function
+ for the register line always defaults back to UDP. (closes issue
+ ASTERISK-21964) Reported by: Doug Bailey Tested by: Doug Bailey
+ Patches: asterisk-21964-set-reg-dialog-transport.diff by Michael
+ L. Young (license 5026)
+
+ * channels/chan_sip.c: Restore Extra Line Break Between Peers When
+ Running AMI Action SIPPeers The commit (r387133) for fixing
+ ASTERISK-21466 accidentally removed an extra line break between
+ the peers returned by the AMI action SIPPeers. This results in
+ some parsers breaking because they expect this extra line break.
+ This patch restores that extra line break. (closes issue
+ ASTERISK-22239) Reported by: Jacek Konieczny Tested by: Jacek
+ Konieczny, Michael L. Young Patches:
+ asterisk-ami_sippeers_separator.patch by Jacek Konieczny (license
+ 6298)
+
+ * UPGRADE.txt: Adding a note to UPGRADE.txt about a change made to
+ res_agi in order to indicate when streaming an audio file fails
+ like it is done in other parts of the code to indicate an error.
+ Note was requested by Paul Belanger:
+ http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
+ (related to issue ASTERISK-21903)
+
+2013-07-22 13:49 +0000 [r394886-395032] Matthew Jordan <mjordan at digium.com>
+
+ * main/asterisk.c: Update copyright year to 2013 in asterisk.c;
+ some whitespace fixes (closes issue ASTERISK-22179) Reported by:
+ Malcolm Davenport
+
+ * funcs/func_channel.c: Clean up documentation This patch cleans up
+ documentation in func_channel for the following items: *
+ rtpsource * secure_signaling * secure_media (closes issue
+ ASTERISK-20969) Reported by: snuffy patches:
+ func_chan-update.diff uploaded by snuffy (License 5024)
+
+ * configs/indications.conf.sample: Provide proper ring tone in
+ indications.conf for Malaysia The ring tone provided in the
+ sample indications.conf was incorrect. This patch modifies the
+ sample ring tone to be what it should: ring =
+ 425/400,0/200,425/400,0/2000 This brings it in line with the tone
+ definition in DAHDI 2.7.0. (zonedata.c) (closes issue
+ ASTERISK-21997) Reported by: Filip Jenicek patches:
+ malaysia_ring.patch uploaded by phill (License 6277)
+
+ * main/http.c: Tolerate presence of RFC2965 Cookie2 header by
+ ignoring it This patch modifies parsing of cookies in Asterisk's
+ http server by doing an explicit comparison of the "Cookie"
+ header instead of looking at the first 6 characters to determine
+ if the header is a cookie header. This avoids parsing "Cookie2"
+ headers and overwriting the previously parsed "Cookie" header.
+ Note that we probably should be appending the cookies in each
+ "Cookie" header to the parsed results; however, while clients can
+ send multiple cookie headers they never really do. While this
+ patch doesn't improve Asterisk's behavior in that regard, it
+ shouldn't make it any worse either. Note that the solution in
+ this patch was pointed out on the issue by the issue reporter,
+ Stuart Henderson. (closes issue ASTERISK-21789) Reported by:
+ Stuart Henderson Tested by: mjordan, Stuart Henderson
+
+ * contrib/realtime/postgresql/realtime.sql: Update PostgreSQL
+ realtime scripts with schema for queue_log table This patch
+ updates the realtime SQL scripts with an entry that will create
+ the queue_log table. This brings the PostgreSQL scripts inline
+ with the MySQL scripts, with respect to what tables they will
+ create. (closes issue ASTERISK-21021) Reported by: Eugene
+ patches: queue_log.sql uploaded by varnav (license 6360)
+
+ * configs/iax.conf.sample: Document connectedline parameter for
+ chan_iax2 The connectedline parameter for a chan_iax2 peer was
+ undocumented. This patch documents the options in the sample
+ configuration file. (closes issue ASTERISK-21953) Reported by:
+ Birger "WIMPy" Harzenetter
+
+2013-07-18 12:51 +0000 [r394640] Michael L. Young <elgueromexicano at gmail.com>
+
+ * res/res_agi.c: Properly indicate failure to open an audio stream
+ in res_agi If there is an error streaming an audio file, the
+ current return status makes it difficult for an AGI script to
+ determine that there was an error with the audio file. This
+ patches changes the result to return -1 and the function returns
+ RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other
+ parts of res_agi, this would appear to be the proper way to
+ handle an error. (closes issue ASTERISK-21903) Reported by: Ariel
+ Wainer Tested by: Ariel Wainer Patches:
+ asterisk-21903-return-stream-res_1.8.diff by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2625/
+
+2013-07-14 01:53 +0000 [r394302] Matthew Jordan <mjordan at digium.com>
+
+ * funcs/func_strings.c: Clarify documentation for function PASSTHRU
+ It is not apparent to the average user that the PASSTHRU function
+ should not be passed as ${PASSTHRU(string)} but just as
+ PASSTHRU(string) to functions which take a variable name and not
+ its contents. This patch clarifies the behavior in the
+ documentation and provides an example. (closes issue
+ ASTERISK-21717) Reported by: Richard Miller patches:
+ func_strings.diff uploaded by Richard Miller (license 5685)
+
+2013-07-11 16:25 +0000 [r394106] Moises Silva <moises.silva at gmail.com>
+
+ * channels/chan_dahdi.c: Fix a longstanding issue with MFC-R2
+ configuration that prevented users from mixing different variants
+ or general MFC-R2 settings within the same E1 line. Most users do
+ not have a problem with this since MFC-R2 lines are usually
+ fractional E1s, or the whole E1 has the same country variant and
+ R2 settings. In Venezuela however is common to have inbound
+ MFC-R2 and outbound DTMF-R2 within the same E1. This fix now
+ properly parses the chan_dahdi.conf file to generate a new openr2
+ context every time a new channel => section is found and the
+ configuration was changed. (closes issue ASTERISK-21117) Reported
+ by: Rafael Angulo Related Elastix issue:
+ http://bugs.elastix.org/view.php?id=1612
+
+2013-07-10 01:41 +0000 [r393928] Russell Bryant <russell at russellbryant.com>
+
+ * configs/sla.conf.sample, include/asterisk/utils.h,
+ apps/app_meetme.c: astobj2-ify the SLA code The SLA code within
+ app_meetme was written before asotbj2 had been merged into
+ Asterisk. Worse, support for reloads did not exist at first and
+ was added later as a bolt-on feature. I knew at the time that
+ reloading was not safe at all while SLA was in use, so the reload
+ would be queued up to execute when the system was idle.
+ Unfortunately, this approach was still prone to errors beyond the
+ fact that this was the only place in Asterisk where configuration
+ was not reloaded instantly when requested. This patch converts
+ various SLA objects to be reference counted objects using
+ astobj2. This allows reloads to be processed while the system is
+ in use. The code ensures that the objects will not disappear
+ while one of the other threads is using them. However, they will
+ be immediately removed from the global trunk and station
+ containers so no new calls will use them if removed from
+ configuration. Review: https://reviewboard.asterisk.org/r/2581/
+
+2013-07-03 23:27 +0000 [r393627] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: Fix segfault reloading
+ chan_dahdi when round robin is used. * Clear round_robin[] in
+ dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo
+ Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621)
+ patch uploaded by rmudgett
+
+2013-06-14 16:14 +0000 [r391778] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_mixmonitor.c: app_mixmonitor: Fix crashes caused by
+ unloading app_mixmonitor Unloading app_mixmonitor while active
+ mixmonitors were running would cause a segfault. This patch fixes
+ that by making it impossible to unload app_mixmonitor while
+ mixmonitors are active. Review:
+ https://reviewboard.asterisk.org/r/2624/
+
+2013-06-12 02:19 +0000 [r391489] Matthew Jordan <mjordan at digium.com>
+
+ * main/loader.c: Fix memory leak while loading priority modules
+ When we load a module with the LOAD_PRIORITY flag, we remove its
+ entry from the load order list. Unfortunately, we don't free the
+ memory associated with entry in the list. This patch corrects
+ that and properly frees the memory for the module in the list.
+
+2013-06-11 08:03 +0000 [r391333] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_iax2.c: IAX2: Transfer Reject: Lock bridgecallno
+ before touching it, refactor 1). When touching the bridgecallno,
+ we need to lock it. 2). Remove magic number '0' and replace with
+ TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2613/
+
+2013-07-15 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.23.0 Released.
+
+2013-07-12 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.23.0-rc2 Released.
+
+ * Properly lock and safely handle a transfer failure in IAX2
+
+ When touching the bridgecallno, we need to lock it - otherwise a
+ race condition can occur. This patch does the proper locking
+ of the bridgecallno before modifying its state.
+
+2013-06-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.23.0-rc1 Released.
+
+2013-06-10 14:15 +0000 [r391215] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, apps/app_queue.c, configs/queues.conf.sample: Add
+ announce-to-first-user option for app_queue In r386792, the
+ ability to play prompts to the first caller in a call queue was
+ added. While this is arguably a bug fix for those who expect the
+ first caller to continue receiving prompts while the agent is
+ dialed, it has the side effect of preventing the first caller
+ from hearing the agent immediately upon bridging. This may not be
+ a problem for those who really want this option, but for those
+ who didn't care whether or not the first caller in queue heard
+ their position, it was an issue. This patch disables the ability
+ for the first caller in the queue to hear prompts and adds a new
+ option, announce-to-first-user, to queues.conf. Those who the
+ behavior can enable it by setting this value to True. Note that
+ if we ever implement the ability to have the prompts be stopped
+ upon bridging, this option can be removed. (closes issue
+ ASTERISK-21782) Reported by: Remi Quezada
+
+2013-06-10 09:30 +0000 [r391062-391143] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * channels/chan_iax2.c: chan_iax2: nativebridge refactor, missed
+ unlock bridgecallno
+
+ * channels/chan_iax2.c: fix bad edit after conflict resolution
+
+ * channels/chan_iax2.c: IAX2: refactor nativebridge transfer remove
+ triple checking of iaxs[fr->callno]->transferring reduce
+ indentation. Reported by: alecdavis Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2602/
+
+ * channels/chan_iax2.c: IAX2: fix race condition with nativebridge
+ transfers. 1). When touching the bridgecallno, we need to lock
+ it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes
+ the lock on the pvt is already held, when iax2_destroy_helper()
+ is called. Thus we need to lock the bridgecallno pvt before we
+ call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When
+ evaluating the state of 'callno->transferring' of the current
+ leg, we can't change it to READY unless the bridgecallno is
+ locked. Why, if we are interrupted by the other call leg before
+ 'transferring = TRANSFER_RELEASED', the interrupt will find that
+ it is READY and that the bridgecallno is also READY so Releases
+ the legs. (closes issue ASTERISK-21409) Reported by: alecdavis
+ Tested by: alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2594/
+
+2013-05-31 08:10 +0000 [r390181] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: reject call attempts when gatekeeper is
+ configured but not registered (closes issue ASTERISK-21800)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch
+ Tested by: Dmitry Melekhov
+
+2013-05-29 20:10 +0000 [r390044] Richard Mudgett <rmudgett at digium.com>
+
+ * main/channel.c: Fix segfault when dealing with chan_agent
+ channels. Check the returned bridged pointer for NULL to avoid a
+ crash. It looks like chan_agent is returning a NULL pointer when
+ it probably should be returning a pointer to the channel the
+ Agent channel is pretending to be. (closes issue ASTERISK-21793)
+ Reported by: Rodrigo P. Telles Patches:
+ jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Rodrigo P. Telles
+
+2013-05-28 17:35 +0000 [r389895] Jonathan Rose <jrose at digium.com>
+
+ * main/slinfactory.c: Fix a memory copying bug in slinfactory which
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