[asterisk-commits] bebuild: tag 10.5.0-rc1 r365140 - /tags/10.5.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 3 07:57:45 CDT 2012
Author: bebuild
Date: Thu May 3 07:57:40 2012
New Revision: 365140
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=365140
Log:
Importing files for 10.5.0-rc1 release.
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tags/10.5.0-rc1/.version (with props)
tags/10.5.0-rc1/ChangeLog (with props)
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+2012-05-03 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.5.0-rc1 Released.
+
+2012-05-02 17:29 +0000 [r365083] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_local.c, /, main/cel.c: Multiple revisions
+ 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03
+ -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race
+ and local channel linkedids This patch has the ;2 channel inherit
+ the linkedid of the ;1 channel and fixes the race condition by no
+ longer scanning the channel list for "other" channels with the
+ same linkedid. Instead, cel.c has an ao2 container of linkedid
+ strings and uses the refcount of the string as a counter of how
+ many channels with the linkedid exist. Not only does this
+ eliminate the race condition, but it also allows us to look up
+ the linkedid by the hashed key instead of traversing the entire
+ channel list. Review: https://reviewboard.asterisk.org/r/1895/
+ ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02
+ May 2012) | 11 lines Don't leak a ref if out of memory and can't
+ link the linkedid If the ao2_link fails, we are most likely out
+ of memory and bad things are going to happen. Before those bad
+ things happen, make sure to clean up the linkedid references.
+ This patch also adds a comment explaining why linkedid can't be
+ passed to both local channel allocations and combines two ao2_ref
+ calls into 1. Review: https://reviewboard.asterisk.org/r/1895/
+ ........ Merged revisions 365006,365068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-02 16:16 +0000 [r365014] Michael L. Young <elgueromexicano at gmail.com>
+
+ * tests/test_security_events.c: Update security events unit tests
+ The security events framework API was changed in Asterisk 10 but
+ the unit tests were not updated at the same time. This patch does
+ the following: * Adds two more security events that were added to
+ the API * Add challenge, received_challenge and received_hash in
+ the inval_password security event unit test (issue
+ ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael
+ L. Young Patches: issue-asterisk-19760-branch10.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/1877/
+
+2012-05-02 02:44 +0000 [r364965] Matthew Jordan <mjordan at digium.com>
+
+ * main/audiohook.c: Only log a failure to get read/write samples
+ from factories if it didn't happen In audiohook_read_frame_both,
+ anytime samples are obtained from the read/write factories a
+ debug statement is logged stating that samples were not obtained
+ from the factories. This statement used to only occur if
+ option_debug was turned on and no samples were obtained; in some
+ refactoring when the option_debug statement was removed, the
+ "else" clause was removed as well. This patch makes it so that
+ those debug log statements only occur if the condition leading up
+ to them actually happened.
+
+2012-05-01 23:14 +0000 [r364903] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
+ (closes issue ASTERISK-19755) Reported by: Gunther Kelleter
+ Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
+ Kelleter ........ Merged revisions 364902 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-01 23:10 +0000 [r364900] Mark Michelson <mmichelson at digium.com>
+
+ * /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
+ error. As it turned out, this wasn't a huge deal. We were calling
+ ast_app_parse_options() for a set of options of which none took
+ arguments. The proper thing to do for this case is to pass NULL
+ for the "args" parameter here. We were instead passing a
+ seemingly-randomly chosen char * from the function. While this
+ would never get written to, you can rest assured things would
+ have gotten bad had new options (which took arguments) been added
+ to func_volume. (closes issue ASTERISK-19656) ........ Merged
+ revisions 364899 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-01 21:50 +0000 [r364845] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_local.c, /: * Fix error path resouce leak in
+ local_request(). * Restructure local_request() to reduce
+ indentation. ........ Merged revisions 364840 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-01 21:44 +0000 [r364842] Jason Parker <jparker at digium.com>
+
+ * main/manager.c, /: Prevent a potential crash when using manager
+ hooks. Found by me while poking at DPMA-127. ........ Merged
+ revisions 364841 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-01 19:07 +0000 [r364787] Kinsey Moore <kmoore at digium.com>
+
+ * /, apps/app_confbridge.c: Play conf-placeintoconf message to the
+ correct channel Correct the code in app_confbridge to play the
+ conf-placeintoconf message to the marked user entering the bridge
+ instead of to the conference while the marked user hears silence.
+ (closes issue ASTERISK-19641) Reported-by: Mark A Walters
+ ........ Merged revisions 364786 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-05-01 18:23 +0000 [r364777] Jonathan Rose <jrose at digium.com>
+
+ * /, main/app.c: Fix bad check in voicemail functions for
+ ast_inboxcount2_func Check looks for ast_inboxcount_func instead
+ of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
+ issue ASTERISK-19718) Reported by: Corey Farrell Patches:
+ ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
+ (license 5909) ........ Merged revisions 364769 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-30 19:42 +0000 [r364707] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Revert improved identities sent in
+ dialog-info NOTIFY requests in r360862 Revision 360862 was
+ intended to improve identities sent in dialog-info NOTIFY
+ requests. Some users reported that hint became broken once this
+ was done. It's not clear exactly what part of the patch has
+ caused this regression, but broken hints are bad. For now, this
+ revision is being reverted so that the next releases of Asterisk
+ do not have bad behavior in them. The original reported issue
+ will have to be fixed differently in the next version of
+ Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-30 16:48 +0000 [r364651] Alexandr Anikin <may at telecom-service.ru>
+
+ * /, addons/ooh323cDriver.c: Fix use freed pointer in return value
+ from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
+ Patches: ASTERISK-19663-ooh323.patch (License #5415) ........
+ Merged revisions 364649 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-30 16:43 +0000 [r364650] Mark Murawki <markm at intellasoft.net>
+
+ * /, main/logger.c: Merged revisions 364635 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
+ 10 lines Sanatize result from bfd_find_nearest_line
+ (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
+ to null resulting in a crash when strrchr(file) runs (closes
+ issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
+ Murawski ........
+
+2012-04-29 19:43 +0000 [r364579] Matthew Jordan <mjordan at digium.com>
+
+ * formats/format_g719.c, formats/format_siren7.c,
+ formats/format_g729.c, formats/format_ilbc.c, /,
+ formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
+ formats/format_pcm.c, formats/format_g723.c,
+ formats/format_h263.c, formats/format_h264.c,
+ formats/format_wav_gsm.c, formats/format_siren14.c,
+ formats/format_gsm.c: Fix error that caused truncate operations
+ to fail Another very inappropriate placement of a ')' (again
+ introduced in r362151) caused the various truncate operations to
+ attempt to truncate the sound file at a position of '0'. (issue
+ ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810)
+ Reported by: colbec ........ Merged revisions 364578 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-29 02:21 +0000 [r364536] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/confbridge/conf_config_parser.c: Fix configuring custom
+ sound_leader_has_left in confbridge.conf The configuration option
+ to specify a custom sound_leader_has_left file for a conference
+ bridge was not being parsed. This patch fixes it so that a custom
+ sound file will now be used. (closes issue ASTERISK-19771)
+ Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young
+ Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak
+ (license 6380) Review: https://reviewboard.asterisk.org/r/1884/
+
+2012-04-27 22:33 +0000 [r364365-364369] Terry Wilson <twilson at digium.com>
+
+ * tests/test_config.c (added): Add missing test_config.c
+
+ * /, main/config.c: Fix ast_parse_arg numeric type range checking
+ and add tests ast_parse_arg wasn't checking for strto* parse
+ errors or limiting the results by the actual range of the numeric
+ types. This patch fixes that and adds unit tests as well. Review:
+ https://reviewboard.asterisk.org/r/1879/ ........ Merged
+ revisions 364340 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-27 21:58 +0000 [r364342] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Don't attempt to make use of the
+ dynamic_exclude_static ACL if DNS lookup fails. (closes issue
+ ASTERISK-18321) Reported by Dan Lukes Patches:
+ ASTERISK-18321.patch by Mark Michelson (license #5049) ........
+ Merged revisions 364341 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-27 19:30 +0000 [r364285] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/time.h, /: Prevent overflow in calculation in
+ ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
+ attempts to calculate the difference, in milliseconds, between
+ two timeval structs, and return the difference in a 64-bit
+ integer. Unfortunately, it assumes that the long tv_sec/tv_usec
+ members in the timeval struct are large enough to hold the
+ calculated values before it returns. On 64-bit machines, this
+ might be the case, as a long may be 64-bits. On 32-bit machines,
+ however, a long may be less (32-bits), in which case, the
+ calculation can overflow. This overflow caused significant
+ problems in MixMonitor, which uses the method to determine if an
+ audio factory, which has not presented audio to an audiohook, is
+ merely late in providing said audio or will never provide audio.
+ In an overflow situation, the audiohook would incorrectly
+ determine that an audio factory that will never provide audio is
+ merely late instead. This led to situations where a MixMonitor
+ never recorded any audio. Note that this happened most frequently
+ when that MixMonitor was started by the ConfBridge application
+ itself, or when the MixMonitor was attached to a Local channel.
+ (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
+ Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
+ #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
+ Murawski Tested by: Michael L. Young Patches:
+ 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
+ (closes issue ASTERISK-19471) Reported by: feyfre Tested by:
+ feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
+ https://reviewboard.asterisk.org/r/1889/ ........ Merged
+ revisions 364277 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-27 18:58 +0000 [r364259] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Allow SIP pvts involved in Replaces
+ transfers to fall out of reference sooner Unref the SIP pvt
+ stored in the refer structure as soon as it is no longer needed
+ so that the pvt and associated file descriptors can be freed
+ sooner. This change makes a reference decrement unnecessary in
+ code that handles SIP BYE/Also transfers which should not touch
+ the reference anyway. (related to issue ASTERISK-19579) ........
+ Merged revisions 364258 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-27 14:44 +0000 [r364204] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: Allow for reloading SRTP crypto keys
+ within the same SIP dialog As a continuation of the patch in
+ r356604, which allowed for the reloading of SRTP keys in
+ re-INVITE transfer scenarios, this patch addresses the more
+ common case where a new key is requested within the context of a
+ current SIP dialog. This can occur, for example, when certain
+ phones request a SIP hold. Previously, once a dialog was
+ associated with an SRTP object, any subsequent attempt to process
+ crypto keys in any SDP offer - either the current one or a new
+ offer in a new SIP request - were ignored. This patch changes
+ this behavior to only ignore subsequent crypto keys within the
+ current SDP offer, but allows future SDP offers to change the
+ keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested
+ by: Thomas Arimont Review:
+ https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
+ 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-27 12:54 +0000 [r364163] Stefan Schmidt <sst at sil.at>
+
+ * res/res_calendar_icalendar.c, res/res_calendar_caldav.c: fix a
+ wrong behavior of alarm timezones in caldav and icalendar when an
+ alarm doesnt use utc. This change uses the same timezone from the
+ start time.
+
+2012-04-26 21:10 +0000 [r364065-364109] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. (With feeling this time.) ........ Merged
+ revisions 364108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix DTMF atxfer running h exten after the
+ wrong bridge ends. When party B does an attended transfer of
+ party A to party C, the attending bridge between party B and C
+ should not be running an h exten when the bridge ends. Running an
+ h exten now sets a softhangup flag to ensure that an AGI will run
+ in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the
+ party B channel for the attending bridge between party B and C.
+ (closes issue AST-870) (closes issue ASTERISK-19717) Reported by:
+ Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
+ Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
+ ........ Merged revisions 364060 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-26 19:30 +0000 [r364047] Terry Wilson <twilson at digium.com>
+
+ * /, main/asterisk.c: Add more constness to the end_buf pointer in
+ the netconsole issue ASTERISK-18308 Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 364046 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-26 13:27 +0000 [r363987] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c: Fix reference leaks involving SIP
+ Replaces transfers The reference held for SIP blind transfers
+ using the Replaces header in an INVITE was never freed on success
+ and also failed to be freed in some error conditions. This caused
+ a file descriptor leak since the RTP structures in use at the
+ time of the transfer were never freed. This reference leak and
+ another relating to subscriptions in the same code path have now
+ been corrected. (closes issue ASTERISK-19579) ........ Merged
+ revisions 363986 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-26 09:46 +0000 [r363935] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * /, channels/chan_sip.c: chan_sip: [general] maxforwards, not
+ checked for a value greater than 255 The peer maxforwards is
+ checked for both '< 1' and '> 255', but the default 'maxforwards'
+ in the [general] section is only checked for '< 1' alecdavis
+ (license 585) Reported by: alecdavis Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1888/ ........ Merged
+ revisions 363934 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-26 03:11 +0000 [r363376-363876] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. (Even better) ........ Merged revisions 363875
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_directed_pickup.c: Update Pickup application
+ documentation. ........ Merged revisions 363788 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_dahdi.c, /, channels/sig_pri.c: Make
+ DAHDISendCallreroutingFacility wait 5 seconds for a reply before
+ disconnecting the call. Some switches may not handle the
+ call-deflection/call-rerouting message if the call is
+ disconnected too soon after being sent. Asteisk was not waiting
+ for any reply before disconnecting the call. * Added a 5 second
+ delay before disconnecting the call to wait for a potential
+ response if the peer does not disconnect first. (closes issue
+ ASTERISK-19708) Reported by: mehdi Shirazi Patches:
+ jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: rmudgett ........ Merged revisions 363730
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+ Clear ISDN channel resetting state if the peer continues to use
+ it. Some ISDN switches occasionally fail to send a RESTART
+ ACKNOWLEDGE in response to a RESTART request. * Made the second
+ SETUP received after sending a RESTART request clear the channel
+ resetting state as if the peer had sent the expected RESTART
+ ACKNOWLEDGE before continuing to process the SETUP. The peer may
+ not be sending the expected RESTART ACKNOWLEDGE. (issue
+ ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
+ jira_ast_815_v1.8.patch (license #5621) patch uploaded by
+ rmudgett (modified) ........ Merged revisions 363687 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Fix recalled party B feature flags for a
+ failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF
+ atxfer to C 3) B hangs up 4) C does not answer 5) B is called
+ back 6) B answers 7) B cannot initiate transfers anymore * Add
+ dial features datastore to recalled party B channel that is a
+ copy of the original party B channel's dial features datastore. *
+ Extracted add_features_datastore() from
+ add_features_datastores(). * Renamed struct ast_dial_features
+ features_caller and features_callee members to my_features and
+ peer_features respectively. These better names eliminate the need
+ for some explanatory comments. * Simplified code accessing the
+ struct ast_dial_features datastore. (closes issue ASTERISK-19383)
+ Reported by: lgfsantos ........ Merged revisions 363428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/features.c: Hangup affected channel in error paths of
+ bridge_call_thread(). ........ Merged revisions 363375 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-23 16:06 +0000 [r363212] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be
+ checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
+ specification does not mandate how these 3 flags must be
+ specified, only that one of the three must be specified in every
+ call. ........ Merged revisions 363209 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-23 14:39 +0000 [r363156] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: AST-2012-004: Fix an error that allows AMI
+ users to run shell commands sans authorization. As detailed in
+ the advisory, AMI users without write authorization for SYSTEM
+ class AMI actions were able to run system commands by going
+ through other AMI commands which did not require that
+ authorization. Specifically, GetVar and Status allowed users to
+ do this by setting their variable/s options to the SHELL or EVAL
+ functions. Also, within 1.8, 10, and trunk there was a similar
+ flaw with the Originate action that allowed users with originate
+ permission to run MixMonitor and supply a shell command in the
+ Data argument. That flaw is fixed in those versions of this
+ patch. (closes issue ASTERISK-17465) Reported By: David Woolley
+ Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
+ (license 6182) ........ Merged revisions 363117 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363141 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-23 14:07 +0000 [r363103-363107] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE
+ handling when no channel owner exists If Asterisk receives a SIP
+ UPDATE request after a call has been terminated and the channel
+ has been destroyed but before the SIP dialog has been destroyed,
+ a condition exists where a connected line update would be
+ attempted on a non-existing channel. This would cause Asterisk to
+ crash. The patch resolves this by first ensuring that the SIP
+ dialog has an owning channel before attempting a connected line
+ update. If an UPDATE request is received and no channel is
+ associated with the dialog, a 481 response is sent. (closes issue
+ ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt
+ Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt
+ Jordan (license 6283) ........ Merged revisions 363106 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_skinny.c: Reference skinny_subchannel object
+ instead of skinny_device for r363103 The check-in to resolve
+ ASTERISK-19592 (r363103) failed to switch to the
+ skinny_subchannel object instead of the skinny_device when
+ attempting to reference the buffer for the keypad digits. This
+ patch fixes that. (issue ASTERISK-19592) Reported by: Russell
+ Bryant
+
+ * /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
+ heap overflow in keypad button handling When handling a keypad
+ button message event, the received digit is placed into a fixed
+ length buffer that acts as a queue. When a new message event is
+ received, the length of that buffer is not checked before placing
+ the new digit on the end of the queue. The situation exists where
+ sufficient keypad button message events would occur that would
+ cause the buffer to be overrun. This patch explicitly checks that
+ there is sufficient room in the buffer before appending a new
+ digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
+ ........ Merged revisions 363100 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+ Merged revisions 363102 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-21 01:45 +0000 [r362998] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, /: Update app_dial M and U option GOTO return
+ value documentation. ........ Merged revisions 362997 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-20 16:47 +0000 [r362918] Michael L. Young <elgueromexicano at gmail.com>
+
+ * main/event.c: Add missing payload type to events API The Security
+ Events Framework API was changed while adding the generation of
+ security events in chan_sip. A payload type and name was missed
+ from being added to struct ie_maps. (closes issue ASTERISK-19759)
+ Reported by: Michael L. Young Patches: issue-asterisk-19759.diff
+ uploaded by Michael L. Young (license 5026)
+
+2012-04-20 16:12 +0000 [r362816-362869] Terry Wilson <twilson at digium.com>
+
+ * /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
+ (closes issue ASTERISK-19758) Reported by: Barry Miller Tested
+ by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
+ (license 5434) ........ Merged revisions 362868 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_speech_utils.c, /: Document Speech* apps hangup on
+ failure and suggest TryExec The Speech API apps return -1 on
+ failure, which will hang up the channel. This may not be
+ desirable behavior for some, but it isn't something that can be
+ changed without breaking people's dialplans or writing an option
+ to all of the Speech apps that does what TryExec already does.
+ This patch documents the hangup behavior of the apps, and
+ suggests TryExec as the solution. (closes issue AST-813) ........
+ Merged revisions 362815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 21:59 +0000 [r362730] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * funcs/func_version.c, /: Fix documentation for
+ ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions
+ 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 21:11 +0000 [r362681] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and
+ trailing backslashes A couple of unit tests did not have have
+ leading or trailing backslashes when setting their test category
+ resulting in a warning message being displayed. Added the
+ backslash where needed. ........ Merged revisions 362680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 21:00 +0000 [r362678] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configs/queues.conf.sample: Update membermacro and membergosub
+ documentation in queues.conf.sample. ........ Merged revisions
+ 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 16:04 +0000 [r362587] Sean Bright <sean at malleable.com>
+
+ * /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when
+ the 'S' command is sent first. If the first command sent from an
+ ExternalIVR client is an 'S' command, we were blindly removing
+ the first element from the play list and deferencing it, even if
+ it was NULL. This corrects that and also locks appropriately in
+ one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
+ ........ Merged revisions 362586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 14:31 +0000 [r362537] Terry Wilson <twilson at digium.com>
+
+ * /, main/asterisk.c: Handle multiple commands per connection via
+ netconsole Asterisk would accept multiple NULL-delimited CLI
+ commands via the netconsole socket, but would occasionally miss a
+ command due to the command not being completely read into the
+ buffer. This patch ensures that any partial commands get moved to
+ the front of the read buffer, appended to, and properly sent.
+ (closes issue ASTERISK-18308) Review:
+ https://reviewboard.asterisk.org/r/1876/ ........ Merged
+ revisions 362536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 02:27 +0000 [r362496] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c,
+ apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
+ addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a
+ variety of potential buffer overflows * chan_mobile: Fixed an
+ overrun where the cind_state buffer (an integer array of size 16)
+ would be overrun due to improper bounds checking. At worst, the
+ buffer can be overrun by a total of 48 bytes (assuming 4-byte
+ integers), which would still leave it within the allocated memory
+ of struct hfp. This would corrupt other elements in that struct
+ but not necessarily cause any further issues. * app_sms: The
+ array imsg is of size 250, while the array (ud) that the data is
+ copied into is of size 160. If the size of the inbound message is
+ greater then 160, up to 90 bytes could be overrun in ud. This
+ would corrupt the user data header (array udh) adjacent to ud. *
+ chan_unistim: A number of invalid memmoves are corrected. These
+ would move data (which may or may not be valid) into the ends of
+ these buffers. * asterisk: ast_console_toggle_loglevel does not
+ check that the console log level being set is less then or equal
+ to the allowed log levels of 32. * format_pref: In
+ ast_codec_pref_prepend, if any occurrence of the specified codec
+ is not found, the value used to index into the array pref->order
+ would be one greater then the maximum size of the array. *
+ jitterbuf: If the element being placed into the jitter buffer
+ lands in the last available slot in the jitter history buffer,
+ the insertion sort attempts to move the last entry in the buffer
+ into one slot past the maximum length of the buffer. Note that
+ this occurred for both the min and max jitter history buffers. *
+ tdd: If a read from fsk_serial returns a character that is
+ greater then 32, an attempt to read past one of the statically
+ defined arrays containing the values that character maps to would
+ occur. * localtime: struct ast_time and tm are not the same size
+ - ast_time is larger, although it contains the elements of tm
+ within it in the same layout. Hence, when using memcpy to copy
+ the contents of tm into ast_time, the size of tm should be used,
+ as opposed to the size of ast_time. * extconf: this treats
+ ast_timing's minmask array as if it had a length of 48, when it
+ has defined the size of the array as 24. pbx.h defines minmask as
+ having a size of 48. (issue ASTERISK-19668) Reported by: Matt
+ Jordan ........ Merged revisions 362485 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-18 16:27 +0000 [r362429] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add
+ ability to ignore layer 1 alarms for BRI PTMP lines. Several
+ telcos bring the BRI PTMP layer 1 down when the line is idle.
+ When layer 1 goes down, Asterisk cannot make outgoing calls.
+ Incoming calls could fail as well because the alarm processing is
+ handled by a different code path than the Q.931 messages. * Add
+ the layer1_presence configuration option to ignore layer 1 alarms
+ when the telco brings layer 1 down. This option can be configured
+ by span while the similar DAHDI driver teignorered=1 option is
+ system wide. This option unlike layer2_persistence does not
+ require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA
+ ABE-2845 ........ Merged revisions 362428 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-17 21:22 +0000 [r362360-362377] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/format_pref.c: Handle case where an unknown format is
+ used to get the preferred codec size In ast_codec_pref_getsize,
+ if an unknown format is passed to the method, no preferred codec
+ will be selected and a negative number will be used to index into
+ the format list. The method now logs an unknown format as a
+ warning, and returns an empty format list. (issue ASTERISK-19655)
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/
+
+ * res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c:
+ Fix places in resources where a negative return value could
+ impact execution This patch addresses a number of modules in
+ resources that did not handle the negative return value from
+ function calls adequately. This includes: * res_agi.c: if the
+ result of the read function is a negative number, indicating some
+ failure, the result would instead be treated as the number of
+ bytes read. This patch now treats negative results in the same
+ manner as an end of file condition, with the exception that it
+ also logs the error code indicated by the return. *
+ res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
+ to srcfd, and instead assigns a negative value, that file
+ descriptor could later be passed to functions that require a
+ valid file descriptor. If spawn_mp3 fails, we now immediately
+ retry instead of continuing in the logic. * res_rtp_asterisk.c:
+ if no codec can be matched between two RTP instances in a peer to
+ peer bridge, we immediately return instead of attempting to use
+ the codec payload type as an index to determine the appropriate
+ negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
+ Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362362 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/manager.c, /, main/asterisk.c: Fix places in main where a
+ negative return value could impact execution This patch addresses
+ a number of modules in main that did not handle the negative
+ return value from function calls adequately, or were not
+ sufficiently clear that the conditions leading to improper
+ handling of the return values could not occur. This includes: *
+ asterisk.c: A negative return value from the read function would
+ be used directly as an index into a buffer. We now check for
+ success of the read function prior to using its result as an
+ index. * manager.c: Check for failures in mkstemp and lseek when
+ handling the temporary file created for processing data returned
+ from a CLI command in action_command. Also check that the result
+ of an lseek is sanitized prior to using it as the size of a
+ memory map to allocate. (issue ASTERISK-19655) Reported by: Matt
+ Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
+ Merged revisions 362359 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-17 20:57 +0000 [r362357] Jonathan Rose <jrose at digium.com>
+
+ * res/res_config_curl.c, res/res_config_pgsql.c,
+ res/res_config_odbc.c, /: Make use of va_args more appropriate to
+ form in various res_config modules plus utils. A number of
+ va_copy operations weren't matched with a corresponding va_end in
+ res_config_odbc. Also, there was a potential for va_end to be
+ invoked twice on the same va_arg in utils, which would mean
+ invoking va_end on an undefined variable... which is bad. va_end
+ is removed from various functions in config_pgsql and config_curl
+ since they aren't making their own copy. The invokers of those
+ functions are responsible for calling va_end on them. (issue
+ ASTERISK-19451) Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/1848/ ........ Merged
+ revisions 362354 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-17 20:56 +0000 [r362305-362356] Matthew Jordan <mjordan at digium.com>
+
+ * /, funcs/func_env.c: Fix places where a negative return from
+ ftello could be used as invalid input In a variety of locations
+ in both reading and writing a file, the result from the C library
+ function ftello is used as input to other functions. For the
+ parameters and functions in question, a negative value is invalid
+ input. This patch checks the return value from the ftello
+ function to determine if we were able to determine the current
+ position in the file stream and, if not, fail gracefully. (issue
+ ASTERISK-19655) Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1863/ ........ Merged
+ revisions 362355 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * formats/format_g719.c, formats/format_siren7.c, /,
+ formats/format_sln.c, formats/format_vox.c, formats/format_wav.c,
+ formats/format_pcm.c, formats/format_wav_gsm.c,
+ formats/format_siren14.c, formats/format_gsm.c: Fix error that
+ caused seek format operations to set max file size to '1' or '0'
+ A very inappropriate placement of a ')' (introduced in r362151)
+ caused the maximum size of a file to be set as the result of a
+ comparison operation, as opposed to the result of the ftello
+ operation. This resulted in seeking being restricted to the
+ beginning of the file, or 1 byte into the file. Thanks to the
+ Asterisk Test Suite for properly freaking out about this on at
+ least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
+ ........ Merged revisions 362304 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-17 14:53 +0000 [r362264] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, channels/chan_sip.c: Turn off warning message when bind
+ address is set to any. When a bind address is set to an ANY
+ address (udpbindport=::), a warning message is displayed stating
+ that "Address remapping activated in sip.conf but we're using
+ IPv6, which doesn't need it. Please remove 'localnet' and/or
+ 'externaddr' settings." But if one is running dual stack, we
+ shouldn't be told to turn those settings off. This patch checks
+ if the bind address is an ANY address or not. The warning message
+ will now only be displayed if the bind address is NOT an ANY
+ address and IPv6 is being used. Also, updated the copyright year.
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