[asterisk-commits] bebuild: tag 1.8.7.0-rc1 r334837 - /tags/1.8.7.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 7 12:25:45 CDT 2011
Author: bebuild
Date: Wed Sep 7 12:25:41 2011
New Revision: 334837
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=334837
Log:
Importing files for 1.8.7.0-rc1 release.
Added:
tags/1.8.7.0-rc1/.lastclean (with props)
tags/1.8.7.0-rc1/.version (with props)
tags/1.8.7.0-rc1/ChangeLog (with props)
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--- tags/1.8.7.0-rc1/ChangeLog (added)
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+2011-09-07 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.7.0-rc1 Released.
+
+2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
+ LIMIT is not portable. Regression from r312212 (closes issue
+ ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
+ Review: https://reviewboard.asterisk.org/r/1415/
+
+2011-09-02 20:59 +0000 [r334296-334355] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_musiconhold.c: MusicOnHold has extra unref which may lead
+ to memory corruption and crash. The problem happens when a call
+ is disconnected and you had started a MOH class that does not use
+ the files mode. If you define REF_DEBUG and recreate the problem,
+ it will announce itself with the following warning: Attempt to
+ unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
+ class is still in a container! * Fixed moh_alloc() and
+ moh_release() functions not handling the state->class reference
+ consistently. (closes issue ASTERISK-18346) Reported by: Mark
+ Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
+ patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
+ Review: https://reviewboard.asterisk.org/r/1404/
+
+ * main/config.c, include/asterisk/config.h: Fix potential memory
+ allocation failure crashes in config.c. * Added required checks
+ to the returned memory allocation pointers to prevent crashes. *
+ Made ast_include_rename() create a replacement ast_variable list
+ node if the new filename is longer than the available space.
+ Fixes potential crash and memory leak. * Factored out
+ ast_variable_move() from ast_variable_update() so
+ ast_include_rename() can also use it when creating a replacement
+ ast_variable list node. * Made the filename stuffed at the end of
+ the struct a minimum allocated size in ast_variable_new() in case
+ ast_include_rename() changes the stored filename. * Constify
+ struct char pointers pointing to strings stuffed at the end of
+ the struct for: ast_variable, cache_file_mtime, and
+ ast_config_map. * Factored out cfmtime_new() to remove inlined
+ code and allow some struct pointers to become const. * Removed
+ the list lock from struct cache_file_mtime that was never used. *
+ Added doxygen comments to several structure elements and better
+ documented what strings are stuffed at the struct end char array.
+ * Reworked ast_config_text_file_save() and set_fn() to handle
+ allocation failure of the include file scratch pad object
+ tracking blank lines. * Made ast_config_text_file_save() fn[]
+ declared with PATH_MAX to ensure it is long enough for any
+ filename with path. Also reduced the number of container fileset
+ buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
+ https://reviewboard.asterisk.org/r/1378/
+
+2011-09-01 17:38 +0000 [r334229-334234] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * main/pbx.c: Remove 1.6 compatibility documentation from 1.8, as
+ it no longer applies.
+
+ * res/res_config_odbc.c: Create a local alias for
+ ast_odbc_clear_cache. As a function pointer, the reference has to
+ be resolved at load time irrespective of the RTLD_LAZY flag.
+ Creating a local alias solves this problem, because the structure
+ is initialized with that local function pointer, while the actual
+ function can remain lazily linked until runtime. The reason why
+ this is important is because we lazily load function references
+ during the module loading process, in order to obtain priority
+ values for each module, ensuring that modules are loaded in the
+ correct order. Previous to this change, when this module was
+ initially loaded, the module loader would emit a symbol
+ resolution error, because of the above requirement. Closes
+ ASTERISK-18399 (reported by Mikael Carlsson, fix suggested by
+ Walter Doekes, patch by me)
+
+2011-08-31 18:50 +0000 [r334156] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: Disable T.38 when we get a invite with image
+ media port set to 0 ASTERISK-17678
+
+2011-08-31 15:57 +0000 [r334009-334012] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: No DAHDI channel available for conference,
+ user introduction disabled. The following error will consistently
+ occur when trying to dial into a MeetMe conference when the
+ server does not have DAHDI hardware installed: app_meetme.c: No
+ DAHDI channel available for conference, user introduction
+ disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
+ correctly during compilation and install of Asterisk/Dahdi,
+ including associated modules, etc., a chan_dahdi.conf
+ configuration file in /etc/asterisk is not created by FreePBX if
+ hardware does not exist, causing MeetMe to be unable to open a
+ DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
+ channel when there is no chan_dahdi.conf file to load. (closes
+ issue ASTERISK-17398) Reported by: Preston Edwards Patches:
+ jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett
+
+ * main/channel.c, channels/chan_agent.c: Call pickup race leaves
+ orphaned channels or crashes. Multiple users attempting to pickup
+ a call that has been forked to multiple extensions either crashes
+ or fails a masquerade with a "bad things may happen" message.
+ This is the scenario that is causing all the grief: 1) Pickup
+ target is selected 2) target is marked as being picked up in
+ ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app
+ dial or queue gets a chance to hang up losing calls and calls
+ ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP
+ YET BY ast_do_pickup() with ast_channel_masquerade(),
+ ast_hangup() completes successfully and the channel is no longer
+ in the channels container. 6) ast_do_pickup() then calls
+ ast_channel_masquerade() to schedule the masquerade on the dead
+ channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the
+ dead channel 8) bad things happen while doing the masquerade and
+ in the process ast_do_masquerade() puts the dead channel back
+ into the channels container 9) The "orphaned" channel is visible
+ in the channels list if a crash does not happen. This patch does
+ the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a
+ successfully hung-up channel and not release the channel lock
+ until that has happened. * Made __ast_channel_masquerade() not
+ setup a masquerade if either channel has AST_FLAG_ZOMBIE set. *
+ Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer
+ work. (closes issue ASTERISK-18222) Reported by: Alec Davis
+ Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes
+ issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by:
+ rmudgett, Alec Davis, irroot, Karsten Wemheuer Review:
+ https://reviewboard.asterisk.org/r/1400/
+
+2011-08-31 15:18 +0000 [r334006] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: Correct an AMI protocol violation with
+ SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n".
+ Since other commands are ended by using \r\n this confuses any
+ interfacing script. (closes issue ASTERISK-17486)
+
+2011-08-30 21:16 +0000 [r333947] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooCalls.c, addons/ooh323c/src/ooh323.c,
+ addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooCalls.h:
+ cleanups in ACF/ARJ GK replies processing fixed long (24 sec)
+ pause if acf/arj proccessed before ast_cond_wait called to wait
+ this
+
+2011-08-29 21:38 +0000 [r333836] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Refresh peer address if DNS unavailable at
+ peer creation If Asterisk starts and no DNS is available,
+ outbound registrations will fail indefinitely. This patch copies
+ the address from the sip_registry struct, which will be updated,
+ to the peer->addr when necessary. If dnsmgr is enabled, the
+ registration fails without the patch because even though the
+ address on the registry is updated via dnsmgr, the address is
+ just copied on the first try. Since we use ast_sockaddr_copy,
+ dnsmgr can't update the address that is copied to the sip_pvt or
+ peers. Closes issue ASTERISK-18000 Review:
+ https://reviewboard.asterisk.org/r/1335/
+
+2011-08-29 21:06 +0000 [r333784-333785] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h: Add some do not hold locks notes to
+ channel.h
+
+ * addons/chan_mobile.c: Fix deadlock potential of
+ chan_mobile.c:mbl_ast_hangup().
+
+2011-08-29 17:11 +0000 [r333630] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Fixed improperly formatted TestEvent AMI
+ message in app_voicemail
+
+2011-08-29 15:55 +0000 [r333569] Jonathan Rose <jrose at digium.com>
+
+ * res/res_jabber.c: Accidental use of variable client->status
+ instead of client->state in from ASTERISK-18078 (issue
+ ASTERISK-18078)
+
+2011-08-28 09:49 +0000 [r333507] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
+ GCC 4.6 detects variables that get assined to, but never used
+ later. Also removes some remmed-out lines that become invalid.
+ (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
+ (License #5035) <tzafrir.cohen at xorcom.com>,
+
+2011-08-26 16:19 +0000 [r333378] Jonathan Rose <jrose at digium.com>
+
+ * res/res_jabber.c: [patch] Buddies are always auto-registered when
+ processing the roster Reporter said autoregister flag was ignored
+ for registering 'buddies' which had a subscription to us.
+ Verified that this was the case and observed how the patch
+ addressed this and made sure it didn't break anything. (closes
+ issue ASTERISK-14233) Reported by: Simon Arlott Patches:
+ asterisk-0015229.patch (license #5756) patch uploaded by Simon
+ Arlott Tested by: Jonathan Rose
+
+2011-08-26 14:36 +0000 [r333339-333354] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: Fixed incorrect pointer copy to structure
+ copy in revision 333339
+
+ * apps/app_voicemail.c: Bug fixes for voicemail user emailsubject /
+ emailbody. This code change fixes a few issues with the voicemail
+ user override of emailbody and emailsubject, including escaping
+ the strings, potential memory leaks, and not overriding the
+ voicemail defaults. Revision 325877 fixed this for
+ ASTERISK-16795, but did not fix it for ASTERISK-16781. A
+ subsequent check-in prevented 325877 from being applied to 10.
+ This check-in resolves both issues, and applies the changes to
+ 1.8, 10, and trunk. (closes issue ASTERISK-16781) Reported by:
+ Sebastien Couture Tested by: mjordan (closes issue
+ ASTERISK-16795) Reported by: mdeneen Tested by: mjordan Review:
+ https://reviewboard.asterisk.org/r/1374
+
+2011-08-25 19:00 +0000 [r333267] Jason Parker <jparker at digium.com>
+
+ * Makefile: Fix for DESTDIR spaces patch.
+
+2011-08-25 18:47 +0000 [r333265] Jonathan Rose <jrose at digium.com>
+
+ * res/res_jabber.c: Segfault when publishing device states via XMPP
+ and not connected When using publishing device state with
+ res_jabber, Asterisk will attempt to send a device state using
+ the unconnected client using iks_send_raw and crash. This patch
+ checks the validity of the connection before attempting to send
+ the device state. (closes issue ASTERISK-18078) Reported by:
+ Michael L. Young Patches:
+ res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
+ patch uploaded by Michael L. Young Tested by: Jonathan Rose
+
+2011-08-25 15:27 +0000 [r333201] Jason Parker <jparker at digium.com>
+
+ * makeopts.in, sounds/Makefile, Makefile, build_tools/mkpkgconfig,
+ configure, configure.ac: Fix installation into directories
+ containing spaces. This also vastly simplifies the logic in
+ sounds/Makefile (Closes issue ASTERISK-18290) Reported by: Paul
+ Belanger Review: https://reviewboard.asterisk.org/r/1379/
+
+2011-08-23 18:14 +0000 [r333010] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: Memory Leak in app_queue The patch that was
+ committed in the 1.6.x versions of Asterisk for ASTERISK-15862
+ actually fixed two issues. One was not applicable to 1.8 but the
+ other is. queue_leak.patch fixes the portion applicable to 1.8.
+ (closes issue ASTERISK-18265) Reported by: Fred Schroeder
+ Patches: queue_leak.patch (license #5049) patch uploaded by
+ mmichelson Tested by: Thomas Arimont
+
+2011-08-23 18:11 +0000 [r333009] Matthew Nicholson <mnicholson at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: default 'sipstorecause' to no We've
+ decided to disable this feature by default in future 1.8
+ versions. This would be an unexpected behavior change for anyone
+ depending on that SIP_CAUSE update in their dialplan. Please
+ refer to the asterisk-dev mailing list more information:
+ http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html
+ (issue AST-580)
+
+2011-08-22 22:11 +0000 [r332939-332945] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c, main/config.c, include/asterisk/config.h:
+ Revert previous commit. Not ready yet.
+
+ * apps/app_queue.c, main/config.c, include/asterisk/config.h:
+ Signed
+
+ * main/config.c: Minor code optimizations. * Simplify
+ ast_category_browse() logic for easier understanding. * Remove
+ dead code in ast_variable_delete() and simplify some of its
+ logic.
+
+2011-08-22 19:41 +0000 [r332876] Paul Belanger <pabelanger at digium.com>
+
+ * channels/chan_gtalk.c: Revert previous commit It seems google is
+ still making changes to the protocol. (issue ASTERISK-18301)
+
+2011-08-22 19:32 +0000 [r332874] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: Reference leaks in app_queue. * Fixed
+ load_realtime_queue() leaking a queue reference when it
+ overwrites q when processing a realtime queue. (issue
+ ASTERISK-18265) * Make join_queue() unreference the queue
+ returned by load_realtime_queue() when it is done with the
+ pointer. The load_realtime_queue() returns a reference to the
+ just loaded realtime queue. * Fixed queues container reference
+ leak in queues_data_provider_get(). * queue_unref() should not
+ return q that was just unreferenced. * Made logic in
+ __queues_show() and queues_data_provider_get() when calling
+ load_realtime_queue() easier to understand.
+
+2011-08-22 18:15 +0000 [r332817] Matthew Jordan <mjordan at digium.com>
+
+ * main/app.c, configs/manager.conf.sample,
+ include/asterisk/manager.h, apps/app_voicemail.c,
+ include/asterisk/test.h, main/manager.c, main/file.c,
+ main/test.c: Review: https://reviewboard.asterisk.org/r/1364/
+ This update adds a new AMI event, TestEvent, which is enabled
+ when the TEST_FRAMEWORK compiler flag is defined. It also adds
+ initial usage of this event to app_voicemail. The TestEvent AMI
+ event is used extensively by the voicemail tests in the Asterisk
+ Test Suite.
+
+2011-08-22 18:14 +0000 [r332759-332816] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_config_pgsql.c, res/res_config_odbc.c: Memory leaks in
+ realtime_multi_xxx() when database access returns error. * Fix
+ realtime_multi_pgsql() configuration memory leak when the
+ database access returns an error. * Fix realtime_multi_odbc()
+ configuration category use after free when the database access
+ returns an error.
+
+ * main/config.c: Memory leak reading realtime database variable
+ list. Calling ast_load_realtime() can leak the last list node if
+ the read list only contains empty variable value items. * Fixed
+ list filter loop in ast_load_realtime() to delete the list node
+ immediately instead of the next time through the loop. The next
+ time through the loop may not happen if the node to delete is the
+ last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
+ Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
+ patch uploaded by rmudgett
+
+2011-08-21 14:31 +0000 [r332699] Paul Belanger <pabelanger at digium.com>
+
+ * channels/chan_gtalk.c: Fix outgoing calls in chan_gtalk (closes
+ issue ASTERISK-18301) Reported by: az1324
+
+2011-08-18 21:26 +0000 [r332559] Terry Wilson <twilson at digium.com>
+
+ * main/netsock2.c: Fix possible error on stringification of
+ IPv4-mapped addrs The FreeBSD netsock2 test has been failing for
+ a while. We were pasing sa->len to getnameinfo instead of
+ sa_tmp->len. ASTERISK-18289
+
+2011-08-18 19:28 +0000 [r332503] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_dahdi.c: CRC4 in "dahdi show status" gives wrong
+ impression to T1 users Change CRC4 to CRC in the output of "dahdi
+ show status" so that it can apply in more situations without
+ confusing users, especially since T1 lines use CRC6 instead of
+ CRC4. (closes issue AST-471)
+
+2011-08-18 14:46 +0000 [r332355-332446] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * build_tools/cflags.xml, build_tools/cflags-devmode.xml: Move
+ BETTER_BACKTRACES out of development mode, as it's useful when
+ DEBUG_THREADS is enabled.
+
+ * makeopts.in, sounds/Makefile, Makefile, agi/Makefile,
+ utils/Makefile, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, Makefile.moddir_rules: Re-add support for spaces in
+ pathnames, including now spaces in DESTDIR. This was initially
+ added to 1.8 prior to release, primarily to support the standard
+ paths on Mac OS X, but was partially reverted recently in
+ Subversion, due to the lack of support for spaces in DESTDIR.
+ This commit restores support for the standard paths on Mac OS X,
+ and also includes support for spaces in DESTDIR. (closes issue
+ ASTERISK-18290) Reported by: pabelanger Review:
+ https://reviewboard.asterisk.org/r/1326/
+
+2011-08-17 17:35 +0000 [r332320] Terry Wilson <twilson at digium.com>
+
+ * res/res_timing_timerfd.c: Don't read from a disarmed or invalid
+ timerfd Numerous isues have been reported for deadlocks that are
+ caused by a blocking read in res_timing_timerfd on a file
+ descriptor that will never be written to. This patch adds some
+ checks to make sure that the timerfd is both valid and armed
+ before calling read(). Should fix: ASTERISK-1842, ASTERISK-18197,
+ ASTERISK-18166, AST-486 AST-495, AST-507 and possibly others.
+
+2011-08-17 15:51 +0000 [r332264] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Outgoing BRI
+ calls fail when using Asterisk 1.8 with HA8, HB8, and B410P
+ cards. France Telecom brings layer 2 and layer 1 down on BRI
+ lines when the line is idle. When layer 1 goes down Asterisk
+ cannot make outgoing calls and the HA8 and HB8 cards also get IRQ
+ misses. The inability to make outgoing calls is because the line
+ is in red alarm and Asterisk will not make calls over a line it
+ considers unavailable. The IRQ misses for the HA8 and HB8 card
+ are because the hardware is switching clock sources from the line
+ which just brought layer 1 down to internal timing. There is a
+ DAHDI option for the B410P card to not tell Asterisk that layer 1
+ went down so Asterisk will allow outgoing calls: "modprobe
+ wcb4xxp teignored=1". There is a similar DAHDI option for the HA8
+ and HB8 cards: "modprobe wctdm24xxp bri_teignored=1".
+ Unfortunately that will not clear up the IRQ misses when the
+ telco brings layer 1 down. * Add layer 2 persistence option to
+ customize the layer 2 behavior on BRI PTMP lines. The new option
+ has three settings: 1) Use libpri default layer 2 setting. 2)
+ Keep layer 2 up. Bring layer 2 back up when the peer brings it
+ down. 3) Leave layer 2 down when the peer brings it down. Layer 2
+ will be brought up as needed for outgoing calls. JIRA AST-598
+
+2011-08-17 14:31 +0000 [r332234] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/chan_sip.c: print a warning instructing the user to
+ disable storesipcause if we process 100 or more scheduler entries
+ at a time AST-580
+
+2011-08-16 20:10 +0000 [r332176] Paul Belanger <pabelanger at digium.com>
+
+ * tests/test_db.c, tests/test_linkedlists.c, tests/test_sched.c,
+ tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
+ tests/test_func_file.c, tests/test_security_events.c,
+ tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
+ tests/test_locale.c, tests/test_acl.c, tests/test_devicestate.c,
+ tests/test_utils.c, tests/test_aoc.c, tests/test_astobj2.c,
+ tests/test_poll.c, tests/test_amihooks.c,
+ tests/test_substitution.c, tests/test_heap.c,
+ tests/test_ast_format_str_reduce.c, tests/test_expr.c,
+ tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
+ tests/test_dlinklists.c, tests/test_event.c: Flag test modules as
+ 'core' Review: https://reviewboard.asterisk.org/r/1369/
+
+2011-08-16 17:38 +0000 [r332118] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: ASTERISK-18067 ASTERISK-15479 - White Space
+ affects mailbox value, multiple MWI subs Before, having multiple
+ subscriptions to mailboxes on a sip peer set via the mailbox
+ setting in sip.conf would only result in updates being sent on
+ whichever mailbox triggered the mwi event. Now all of them get
+ counted regardless. Also fixes a bug involving parsing of the
+ mailbox option in sip.conf so that trailing and leading spaces
+ before/after commas are trimmed. (closes issue ASTERISK-18067)
+ Reported by: aragon (closes issue ASTERISK-15479) Reported by:
+ Ben Winslow Patches:
+ chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
+ patch uploaded by Ben Winslow
+
+2011-08-16 16:31 +0000 [r332100] Richard Mudgett <rmudgett at digium.com>
+
+ * CHANGES, configs/features.conf.sample, main/asterisk.c,
+ main/features.c: Fix multiple parking issues. JIRA ASTERISK-17183
+ Multi-parkinglot directs calls to wrong parkinglot. JIRA
+ ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
+ ParkedCall() with no extension should pickup first available call
+ and does not. JIRA AST-576 Issues with parking lots * Removed
+ searching for parking lots by extension. Parking lots can only be
+ found by the parking lot name since parking lot access extensions
+ and spaces are not guaranteed to be unique. * Added
+ parking_lot_name option to the Park and ParkedCall applications.
+ Updated documentation for Park and ParkedCall applications. * Add
+ parkext_exclusive configuration option to make parking entry
+ extensions specify which parking lot they access. (closes issue
+ ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
+ David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
+ Quezada (closes issue ASTERISK-17430) Reported by: Philippe
+ Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
+ AST-624 'next' setting for findslot does nothing * Reimplemented
+ since findslot feature option broken by -r114655. (closes issue
+ ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
+ JIRA ASTERISK-15792 Dialplan continues execution after transfer
+ to park. This happens for DTMF attended transfer, DTMF blind
+ transfer, and DTMF one-touch-parking if the party initiating
+ these features also initiated the call. * Fixed the return code
+ from the affected builtin features when parking a call. (closes
+ issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
+ rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
+ the expected call when picking up a parked call. This is mostly a
+ documentation problem. However, the option is not reset to the
+ default when features.conf is reloaded. * Updated
+ features.conf.sample documentation for courtesytone and
+ parkedplay options. * Reset the parkedplay option to default when
+ features.conf is reloaded. JIRA AST-615 AMI Park action followed
+ by features reload results in orphaned channels in parking lot. *
+ Reloading features.conf will not touch parking lots that have
+ calls still parked in them. Reload again at a later time. Misc
+ additional fixes: * Added unit test for parking lot dialplan
+ usage checking. * Made update connected line when a parked call
+ is retrieved from a parking lot. * Made retrieved parked call
+ stop ringing or MOH depending upon how the call was waiting in
+ the parking lot. * Made CLI "features show" indicate if the
+ parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
+ variable to allow dynamic parking lots to specify the parking lot
+ access extension. * Made AMI ParkedCalls action ParkedCall events
+ have a Parkinglot header. * Made AMI ParkedCalls action
+ ParkedCallsComplete event have a Total header. * Fixed potential
+ deadlock from AMI Park action holding channel locks while calling
+ masq_park_call(). * Fixed several places where ast_strdupa() were
+ used inside of loops. (Mostly fixed by refactoring the loop body
+ into its own function.) * Fixed copy_parkinglot() copying too
+ much from the source parking lot. Extracted the parking lot
+ configuration settings into struct parkinglot_cfg. * Refactored
+ courtesytone playing code to put the channel not playing the tone
+ in autoservice. * Fix when pbx-parkingfailed is played that the
+ other channel is put in autoservice if it exists. * Fixed
+ parkinglot reference leak in parked_call_exec() error paths. *
+ Fixed parkinglot_unref() use of parkinglot after it was unreffed.
+ * Made destroy the struct ast_parkinglot parkings lock when done.
+ * Refactored the features.conf parking lot configuration code to
+ eliminate redundancy. * Fixed feature reload to better protect
+ parking lots. * Fixed parking lot container reference leak in
+ handle_parkedcalls(). * Fixed the total count in
+ handle_parkedcalls(). Review:
+ https://reviewboard.asterisk.org/r/1358/
+
+2011-08-16 15:06 +0000 [r332021-332026] Matthew Nicholson <mnicholson at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: use
+ DEFAULT_STORE_SIP_CAUSE to set the default value for the
+ 'storesipcause' option AST-580
+
+ * configs/sip.conf.sample, CHANGES, channels/chan_sip.c: Added the
+ 'storesipcause' option to sip.conf to allow the user to disable
+ the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having
+ chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
+ significant performance penalty because of the usage of the
+ MASTER_CHANNEL() dialplan function. AST-580
+
+2011-08-15 17:24 +0000 [r331955] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix some minor chan_dahdi config load
+ issues. * Address chan_dahdi.conf dahdichan option todo item
+ about needing line number. * Make ignore_failed_channels option
+ also apply to dahdichan option. * Don't attempt to create a
+ default pseudo channel if the chan_dahdi.conf channel/channels
+ option is not allowed. * Add a similar check for dahdichan in
+ normal chan_dahdi.conf sections as is done in users.conf.
+
+2011-08-15 15:21 +0000 [r331886] Paul Belanger <pabelanger at digium.com>
+
+ * main/rtp_engine.c: Fix noisy message when briding channels
+ (closes issue ASTERISK-18270) Reported by: Federico Alves
+
+2011-08-15 15:12 +0000 [r331867] David Vossel <dvossel at digium.com>
+
+ * channels/chan_sip.c: Fixes locking inversion issues present in
+ the handling of the sip REFER method. (closes issue
+ ASTERISK-18082) Reported by: James Van Vleet
+
+2011-08-12 19:01 +0000 [r331774] Matthew Nicholson <mnicholson at digium.com>
+
+ * apps/app_queue.c: Unlock the channel before calling update_queue.
+ Holding the channel lock when calling update_queue which attempts
+ to lock the queue lock can cause a deadlock. This deadlock
+ involves the following chain: 1. hold chan lock -> wait queue
+ lock 2. hold queue lock -> wait agent list lock 3. hold agent
+ list lock -> wait chan list lock 4. hold chan list lock -> wait
+ chan lock
+
+2011-08-12 18:58 +0000 [r331714-331771] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Suppress warning message when using
+ DAHDITransfer or DAHDIHangup. * The fake event should only be
+ processed by the channel that currently owns the private and not
+ the associated call waiting or 3-way channel. JIRA AST-620 JIRA
+ SWP-3616
+
+ * channels/chan_dahdi.c: AMI actions DAHDIHangup and DAHDITransfer
+ have no effect. The AMI actions DAHDIHangup and DAHDITransfer
+ have no effect on a DAHDI channel. These two AMI actions are
+ highly specialized to analog channels and appear to make the
+ channel behave like a jack port for headsets. * Made the faked
+ DAHDI event get processed before a normal media stream read in
+ dahdi_read() instead of trying to trigger an exception read by
+ setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
+ long ago that changed how AST_FLAG_EXCEPTION is processed in the
+ core. Unfortunately, the faked DAHDI events no longer worked when
+ that happened. * Updated the DAHDI AMI action documentation for
+ the following actions: DAHDITransfer, DAHDIHangup,
+ DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
+ DAHDIRestart. * Made use sscanf() instead of atoi() for better
+ error checking of the DAHDIChannel header string. JIRA AST-620
+ JIRA SWP-3616
+
+2011-08-12 16:30 +0000 [r331658] Terry Wilson <twilson at digium.com>
+
+ * tests/test_netsock2.c: Fix netsock2 multiple zero-expansion test
+ Remove erroneous single bracket.
+
+2011-08-12 16:20 +0000 [r331649] Kinsey Moore <kmoore at digium.com>
+
+ * main/logger.c: Logger does not warn of failure to open logging
+ channels Currently, logger only prints an error message to stderr
+ when it fails to open a logger channel where many users will not
+ see it because the logger lock is held. The alternative provided
+ by this patch is to log the error to all attached consoles in the
+ hopes that it will be easier to see. Additionally, this patch
+ prevents the failed logger channel from being added to the list
+ where it would silently fail on each call to the Asterisk logger.
+ (closes issue ASTERISK-16231) Review:
+ https://reviewboard.asterisk.org/r/1338
+
+2011-08-12 15:49 +0000 [r331635] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_dial.c, apps/app_meetme.c: Fixes 32bit compilation
+ warnings brought on by 331634 in app_dial and app_meetme
+
+2011-08-11 21:46 +0000 [r331578] Jason Parker <jparker at digium.com>
+
+ * apps/app_dial.c, apps/app_meetme.c: Use proper values for 64-bit
+ option flags. Also, reusing bits es no bueno, so change the value
+ of a duplicate. (issue ASTERISK-18239)
+
+2011-08-11 21:39 +0000 [r331575] Richard Mudgett <rmudgett at digium.com>
+
+ * funcs/func_shell.c: Segfault in shell_helper in func_shell.c. The
+ return value of popen() was not checked for failure to open.
+ (closes issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael
+ Myles Tested by: rmudgett
+
+2011-08-10 22:23 +0000 [r331517] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: SIP Notify via AMI or CLI leaks SIP PVTs Any
+ SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2.
+ Removing the additional ref just before the invite and adding an
+ unref following it corrects the issue as seen via REF_DEBUG. The
+ unref existed in a distant revision and it appears as though the
+ wrong ref operation was removed. (closes issue ASTERISK-18091)
+ Review: https://reviewboard.asterisk.org/r/1332/
+
+2011-08-10 20:29 +0000 [r331461] Richard Mudgett <rmudgett at digium.com>
+
+ * main/logger.c: Output of queue log not started until logger
+ reloaded. ASTERISK-15863 caused a regression with queue logging.
+ The output of the queue log is not started until the logger
+ configuration is reloaded. * Queue log initialization is
+ completely delayed until the first message is posted to the queue
+ log system. Including the initial opening of the queue log file.
+ * Fixed rotate_file() ROTATE strategy to give the file just
+ rotated out to the configured exec function after rotate. Just
+ like the other strategies. * Fixed logger reload to always post
+ the queue reload entry instead of just if there is a queue log
+ file. * Refactored some code to eliminate some redundancy and to
+ reduce stack utilization. (closes issue ASTERISK-17036) JIRA
+ SWP-2952 Reported by: Juan Carlos Valero Patches:
+ jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett (closes issue ASTERISK-18208)
+ Reported by: Christian Pinedo Review:
+ https://reviewboard.asterisk.org/r/1333/
+
+2011-08-31 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.6.0 Released.
+
+2011-08-25 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.6.0-rc3 Released.
+
+ ------------------------------------------------------------------------
+ r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | 8 lines
+
+ Fix installation into directories containing spaces.
+
+ This also vastly simplifies the logic in sounds/Makefile
+
+ (Closes issue ASTERISK-18290)
+ Reported by: Paul Belanger
+ Review: https://reviewboard.asterisk.org/r/1379/
+ ------------------------------------------------------------------------
+
+2011-08-22 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.6.0-rc2 Released.
+
+ ------------------------------------------------------------------------
+ r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) | 9 lines
+
+ Segfault in shell_helper in func_shell.c.
+
+ The return value of popen() was not checked for failure to open.
+
+ (closes issue ASTERISK-18109)
+ JIRA SWP-3633
+ Reported by: Michael Myles
+ Tested by: rmudgett
+ ------------------------------------------------------------------------
+ r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 13 lines
+
+ Re-add support for spaces in pathnames, including now spaces in DESTDIR.
+
+ This was initially added to 1.8 prior to release, primarily to support the
+ standard paths on Mac OS X, but was partially reverted recently in Subversion,
+ due to the lack of support for spaces in DESTDIR. This commit restores support
+ for the standard paths on Mac OS X, and also includes support for spaces in
+ DESTDIR.
+
+ (closes issue ASTERISK-18290)
+ Reported by: pabelanger
+
+ Review: https://reviewboard.asterisk.org/r/1326/
+ ------------------------------------------------------------------------
+ r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) | 7 lines
+
+ Fix possible error on stringification of IPv4-mapped addrs
+
+ The FreeBSD netsock2 test has been failing for a while. We were
+ pasing sa->len to getnameinfo instead of sa_tmp->len.
+
+ ASTERISK-18289
+ ------------------------------------------------------------------------
+
+2011-08-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.6.0-rc1 Released.
+
+2011-08-10 13:47 +0000 [r331315] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c: AMI action ModuleReload returns Error if Module:
+ missing or empty An empty string was not being checked for
+ properly causing identification of the module to be reloaded to
+ fail and return an Error with message "No such module." (closes
+ issue AST-616)
+
+2011-08-09 22:12 +0000 [r331248] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c, apps/app_parkandannounce.c, main/pbx.c,
+ channels/chan_sip.c, main/features.c: Misc minor items found in
+ code. * Add some reentrancy protection in pbx.c when creating the
+ contexts_table hash table. * Fix inverted test in chan_sip.c
+ conditional code. * Fix uninitialized variable and use of the
+ wrong variable in chan_iax2.c. * Fix test of return value in
+ app_parkandannounce.c. Explicitly testing for -1 is bad if the
+ function does not actually return that value when it fails. *
+ Fixup some comments and add some curly braces in features.c.
+
+2011-08-09 16:13 +0000 [r331146] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c: move ast_cond_signal for admitted call
+ after all data filled/freed clear all log channels by pointed
+ number not only first free allocated callToken in ooh323_answer
+
+2011-08-09 15:58 +0000 [r331142] Jason Parker <jparker at digium.com>
+
+ * doc/asterisk.8: Regenerate asterisk man page from sgml.
+
+2011-08-08 20:52 +0000 [r331038] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_musiconhold.c: In-queue MOH stops after a periodic
+ announcement If the seek value is past the end of file when
+ resuming G.722 MOH, MOH will cease to function for the duration
+ of the MOH session through all starts and stops until saved state
+ is cleared. Adjusting the code to guarantee a single valid read
+ (which is already assumed) fixes the bug. (closes issue
+ ASTERISK-18077) Review: https://reviewboard.asterisk.org/r/1328/
+ Tested-by: Jonathan Rose <jrose at digium.com>
+
+2011-08-04 20:29 +0000 [r330843] Terry Wilson <twilson at digium.com>
+
[... 33002 lines stripped ...]
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