[svn-commits] bebuild: tag 1.8.26.0-rc1 r405528 - /tags/1.8.26.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Jan 14 13:33:21 CST 2014
Author: bebuild
Date: Tue Jan 14 13:33:18 2014
New Revision: 405528
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=405528
Log:
Importing files for 1.8.26.0-rc1 release.
Added:
tags/1.8.26.0-rc1/.lastclean (with props)
tags/1.8.26.0-rc1/.version (with props)
tags/1.8.26.0-rc1/ChangeLog (with props)
Added: tags/1.8.26.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/.lastclean?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/.lastclean (added)
+++ tags/1.8.26.0-rc1/.lastclean Tue Jan 14 13:33:18 2014
@@ -1,0 +1,1 @@
+40
Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.26.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.26.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/.version?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/.version (added)
+++ tags/1.8.26.0-rc1/.version Tue Jan 14 13:33:18 2014
@@ -1,0 +1,1 @@
+1.8.26.0-rc1
Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.26.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.26.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.26.0-rc1/ChangeLog?view=auto&rev=405528
==============================================================================
--- tags/1.8.26.0-rc1/ChangeLog (added)
+++ tags/1.8.26.0-rc1/ChangeLog Tue Jan 14 13:33:18 2014
@@ -1,0 +1,47186 @@
+2014-01-14 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.26.0-rc1 Released.
+
+2014-01-14 18:35 +0000 [r405433-405486] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * channels/chan_sip.c: chan_sip: No BYE message sent after INVITE
+ with Replaces Setting channel state DOWN is an unnecessary step
+ that was only being done in handle_invite_replaces(). This
+ changes that by removing the call and reducing locking. (closes
+ issue ASTERISK-23010) Reported by: Ryan Tilton Review:
+ https://reviewboard.asterisk.org/r/3116/
+
+ * channels/chan_sip.c: chan_sip: fix Local From tag on outbound
+ register regression In ASTERISK-12117, an improvement to insure
+ consistant local from tags on outbound registrations resulted in
+ an undesirable behavior - caused by leftover unexpired sip_pvt
+ dialogs (with the previous cseq number), resulting in many
+ uncessary REGISTER requests. Instead of significant rework of
+ transmit_register(), this change deletes the dialogs after a 200
+ OK response indiciating a successful registration, keeping the
+ old dialogs from interfering with normal operation. (closes issue
+ ASTERISK-22946) Reported by: Stephan Eisvogel Review:
+ https://reviewboard.asterisk.org/r/3109/
+
+2014-01-14 15:31 +0000 [r405379] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Hangup transferer/transferee when
+ transfer to Parking fails When performing a SIP transfer to a
+ Park extension, if the Park fails, chan_sip will currently not
+ hang up either the transferer or the transfer target. This
+ results in the channels being orphaned with no thread to service
+ frames, resulting in stuck channels. This patch immediately hangs
+ up the two channels if a Park fails. (closes issue
+ ASTERISK-22834) Reported by: rsw686 (closes issue ASTERISK-23047)
+ Reported by: Tommy Thompson Review:
+ https://reviewboard.asterisk.org/r/3107
+
+2014-01-09 14:11 +0000 [r405160] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_dumpchan.c: "Minimun" typo.
+
+2014-01-08 16:00 +0000 [r405033-405090] Kinsey Moore <kmoore at digium.com>
+
+ * configure, configure.ac, pbx/pbx_lua.c: pbx_lua: Add support for
+ Lua 5.2 This adds support for Lua 5.2 in pbx_lua which is
+ available on newer operating systems. (closes issue
+ ASTERISK-23011) Review: https://reviewboard.asterisk.org/r/3075/
+ Reported by: George Joseph Patch by: George Joseph
+
+ * UPGRADE.txt: UPGRADE: Add a note about non-functionality Add a
+ note that the "retry on 403 response to REGISTER" for chan_sip is
+ non-functional in the versions in which it was first introduced.
+
+ * channels/chan_sip.c: Add the missing part of r400140 When the
+ patch to add retry-on-forbidden-response was committed, part of
+ the patch for chan_sip was not committed which caused the feature
+ to be entirely nonfunctional. This corrects the code in question.
+ (closes issue ASTERISK-17138) Review:
+ https://reviewboard.asterisk.org/r/2874
+
+2014-01-06 17:31 +0000 [r404951] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * funcs/func_strings.c: func_strings: fix for memmove patch test In
+ r404674 the AST_TEST_DEFINE(test_REPLACE) test was added that
+ made use of a function that doesn't exist in 1.8. This fixes that
+ by reverting to directly accessing chan varshead. Reported by:
+ Tzafrir Cohen (issue ASTERISK-22910)
+
+2014-01-03 22:06 +0000 [r404861] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * main/asterisk.c: asterisk.c: supress live_dangerously warning on
+ rasterisk Even since the fixes of AST-2013-007, Asterisk prints
+ the following warning on startup if the user decided to live
+ dangerously: Privilege escalation protection disabled! See
+ https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This
+ message is intended for the logs and interactive startup. No need
+ for it to appear on a remote console. This commit removes it from
+ there. (closes issue ASTERISK-23084) Review:
+ https://reviewboard.asterisk.org/r/3101/
+
+2014-01-03 21:57 +0000 [r404742-404857] Kevin Harwell <kharwell at digium.com>
+
+ * cel/cel_pgsql.c: cel_pgsql: module not correctly reloading Upon
+ reload the module unconditionally "unloaded" the module (freeing
+ memory and setting pointers to NULL) and then when attempting a
+ "load" if the config file had not changed then nothing would be
+ reinitialized. By moving the "unload" to occur conditionally
+ (reload only) after an attempted configuration load, but before
+ module "loading" alleviates the issue. The module now
+ loads/unloads/reloads correctly. (closes issue ASTERISK-22871)
+ Reported by: Matteo
+
+ * channels/chan_dahdi.c: chan_dahdi: dahdi show channels slices PRI
+ channel dnid on output dahdi show channels output slices the
+ callerid (which is dnid copied over on PRI channels). If the
+ channel naming structures look like: 'DAHDI/i1/1408409XXXX-6'
+ then the output slices 1408409XXXX down to 1408409XXX. This patch
+ just opens it up to 15 chars so you can see the whole thing.
+ (closes issue ASTERISK-22918) Reported by: outtolunc Patches:
+ svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc
+ (license 5198)
+
+ * apps/app_meetme.c, channels/chan_unistim.c: chan_unistim.c,
+ app_meetme: compiler warnings Fixed a couple of compiler warnings
+ (errors in 'dev-mode') given by gcc version 4.8.1. The one in
+ app_meetme involved the 'sizeof-pointer-memaccess' (see:
+ http://gcc.gnu.org/gcc-4.8/porting_to.html) warning. The one in
+ chan_unistim was issuing an array out of bounds message. Fixed
+ both so they would no longer issue warnings and can compile again
+ in 'dev-mode'. Review: https://reviewboard.asterisk.org/r/3098/
+
+2014-01-02 19:32 +0000 [r404674] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * funcs/func_strings.c: func_strings: use memmove to prevent
+ overlapping memory on strcpy When calling REPLACE() with an empty
+ replace-char argument, strcpy is used to overwrite the the
+ matching <find-char>. However as the src and dest arguments to
+ strcpy must not overlap, it causes other parts of the string to
+ be overwritten with adjacent characters and the result is
+ mangled. Patch replaces call to strcpy with memmove and adds a
+ test suite case for REPLACE. (closes issue ASTERISK-22910)
+ Reported by: Gareth Palmer Review:
+ https://reviewboard.asterisk.org/r/3083/ Patches:
+ func_strings.patch uploaded by Gareth Palmer (license 5169)
+
+2013-12-31 21:25 +0000 [r404603] Kevin Harwell <kharwell at digium.com>
+
+ * cel/cel_pgsql.c: cel_pgsql: deadlock on unload and
+ core_event_dispatcher A deadlock can happen between a thread
+ unloading or reloading the cel_pgsql module and the
+ core_event_dispatcher taskprocessor thread. Description of what
+ is happening: Thread 1 (for example, a netconsole thread): a
+ "module reload cel_pgsql" is launched the thread enter the
+ "my_unload_module" function (cel_pgsql.c) the thread acquire the
+ write lock on psql_columns the thread enter the
+ "ast_event_unsubscribe" function (event.c) the thread try to
+ acquire the write lock on ast_event_subs[sub->type] Thread 2
+ (core_event_dispatcher taskprocessor thread): the taskprocessor
+ pop a CEL event the thread enter the "handle_event" function
+ (event.c) the thread acquire the read lock on
+ ast_event_subs[sub->type] the thread callback the "pgsql_log"
+ function (cel_pgsql.c), since it's a subscriber of CEL events the
+ thread try to acquire a read lock on psql_columns (closes issue
+ ASTERISK-22854) Reported by: Etienne Lessard Patches:
+ cel_pgsql_fix_deadlock_event.patch uploaded by hexanol (license
+ 6394)
+
+2013-12-20 21:12 +0000 [r404456] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/say.c: say.c: correct time for polish In
+ ast_say_date_with_format_pl(), change ast_say_number() to use
+ tm_sec instead of tm_mn. (closes issue ASTERISK-22856) Reported
+ by: Robert Mordec Review:
+ https://reviewboard.asterisk.org/r/3082/ Patches: say.c.patch
+ uploaded by veilen (license 6555)
+
+2013-12-18 19:47 +0000 [r404212] Richard Mudgett <rmudgett at digium.com>
+
+ * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/memheap.c,
+ addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
+ addons/ooh323c/src/perutil.c, addons/ooh323cDriver.c,
+ addons/ooh323c/src/ooSocket.c: ooh323c: Fix gcc 4.6.3 compiler
+ warnings.
+
+2013-12-18 11:58 +0000 [r404135] Joshua Colp <jcolp at digium.com>
+
+ * res/res_calendar.c: res_calendar: Protect channel when adding
+ datastore. This change adds a missing channel lock when adding a
+ datastore to a channel.
+
+2013-12-18 00:27 +0000 [r404044-404081] Rusty Newton <rnewton at digium.com>
+
+ * funcs/func_strings.c: func_strings: Documentation fix for QUOTE()
+ Example output was inaccurate. (issue ASTERISK-22970) (closes
+ issue ASTERISK-22970) Reported by: Gareth Palmer Patches:
+ func_strings.patch uploaded by Gareth Palmer (license 5169)
+
+ * channels/chan_iax2.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+ include/asterisk/test.h, main/channel.c: Several components:
+ fixing Typos in comments and code, "avaliable" instead of
+ "available" (issue ASTERISK-23021) (closes issue ASTERISK-23021)
+ Reported by: Jeremy Lainé Tested by: Rusty Newton Patches:
+ available.patch uploaded by Jeremy Lainé (license 6561)
+
+2013-12-16 16:36 +0000 [r403913] David M. Lee <dlee at digium.com>
+
+ * include/asterisk/pbx.h, main/asterisk.c, funcs/func_realtime.c,
+ main/pbx.c, main/tcptls.c, funcs/func_db.c,
+ README-SERIOUSLY.bestpractices.txt, configs/asterisk.conf.sample,
+ funcs/func_shell.c, funcs/func_env.c, funcs/func_lock.c,
+ UPGRADE.txt: security: Inhibit execution of privilege escalating
+ functions This patch allows individual dialplan functions to be
+ marked as 'dangerous', to inhibit their execution from external
+ sources. A 'dangerous' function is one which results in a
+ privilege escalation. For example, if one were to read the
+ channel variable SHELL(rm -rf /) Bad Things(TM) could happen;
+ even if the external source has only read permissions. Execution
+ from external sources may be enabled by setting
+ 'live_dangerously' to 'yes' in the [options] section of
+ asterisk.conf. Although doing so is not recommended. (closes
+ issue ASTERISK-22905) Review:
+ http://reviewboard.digium.internal/r/432/
+
+2013-12-16 15:53 +0000 [r403853-403862] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/pbx.c: pbx.c: put copy of ast_exten.data on stack to prevent
+ memory corruption During dialplan execution in
+ pbx_extension_helper(), the contexts global read lock prevents
+ link list corruption, but was released with a pointer to the
+ ast_exten and data later used in variable substitution. Instead,
+ this patch removes pbx_substitute_variables() and locates a copy
+ of the ast_exten data on the stack before releasing the lock,
+ where ast_exten could get free'd by another thread performing a
+ module reload. (issue AST-1179) Reported by: Thomas Arimont
+ (issue AST-1246) Reported by: Alexander Hömig Review:
+ https://reviewboard.asterisk.org/r/3055/
+
+ * apps/app_sms.c: app_sms: BufferOverflow when receiving odd length
+ 16 bit message This patch prevents an infinite loop overwriting
+ memory when a message is received into the unpacksms16()
+ function, where the length of the message is an odd number of
+ bytes. (closes issue ASTERISK-22590) Reported by: Jan Juergens
+ Tested by: Jan Juergens
+
+2013-12-11 19:11 +0000 [r403634] Russell Bryant <russell at russellbryant.com>
+
+ * channels/chan_sip.c: Reset peer outboundproxy on sip.conf reload
+ If you set a peer's outboundproxy and then removed it from the
+ config, this would not get picked up in a config reload. This
+ patch fixes that by resetting it in set_peer_defaults(). Closes
+ ASTERISK-19454 Review: https://reviewboard.asterisk.org/r/3065/
+
+2013-12-09 03:10 +0000 [r403449] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: Always init T.38 session
+ to avoid crashes during state change Prior to this patch,
+ res_fax_spandsp was conservative with how it initialized the
+ spandsp T.38 context. It would only initialize it if the driver
+ thought the current state was a T.38 fax. While this works fine
+ in nominal situations, in certain off nominal situations,
+ res_fax_spandsp can believe that a T.38 fax will not occur when
+ in fact one has started. In particular, this was discovered when
+ res_fax would fall back to audio after timing out on a T.38
+ upgrade. The SIP channel driver would continue to retry the
+ re-INVITE and - if the remote end responded after res_fax timed
+ out with a 200 OK - a T.38 frame would be delivered to the
+ res_fax stack when it no longer expected it. As it turns out,
+ there does not appear to be any downside to always initializing
+ the T.38 context, other than the actual memory allocation. Since
+ that avoids this off nominal situation (and others which are
+ equally likely hard to predict), this is the safest way to avoid
+ this problem. Much thanks to Torrey as well for providing a
+ scenario that reproduces this issue. (closes issue
+ ASTERISK-21242) Reported by: Ashley Winters Tested by: Torrey
+ Searle patches: always-init-t38.patch uploaded by awinters
+ (License 6477) A_PARTY.xml uploaded by tsearle (License 5334)
+
+2013-11-22 17:10 +0000 [r403014] Joshua Colp <jcolp at digium.com>
+
+ * main/translate.c: translate: Move freeing of frame to after it is
+ used. When translating from one format to another it is possible
+ to inform the translation function that the source frame should
+ be freed. This was previously done immediately but shortly
+ afterwards the frame that was freed was accessed and used again.
+ This change moves code around a bit so that the frame is now
+ freed after it has been completely used. (closes issue
+ ASTERISK-22788) Reported by: Corey Farrell Patches:
+ translate-access-after-free-11up.patch uploaded by coreyfarrell
+ (license 5909) translate-access-after-free-1.8.patch uploaded by
+ coreyfarrell (license 5909)
+
+2013-11-12 14:55 +0000 [r402645-402708] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: Fix crash during caller ID
+ read Asterisk will sometimes core dump during caller id read on
+ analog channels due to a negative return value from the read() in
+ my_get_callerid that slips through as a negative length argument
+ to callerid_feed() if the errno returned by DAHDI is ELAST. This
+ change ensures that the negative return is treated properly even
+ when it is ELAST. (closes issue ASTERISK-22746) Reported by:
+ Michael Walton Patches: chan_dahdi_cid_crash_fix.r401410.patch
+ uploaded by Michael Walton (License 6502)
+
+ * apps/app_queue.c: app_queue: Honor penalty limits of 0 In the
+ current app_queue code from 1.8 up to trunk the upper and lower
+ penalties can be set to 0 but the value is interpreted to be
+ disabled instead of actually setting limits. This is especially
+ evident if min and max limits are set to 0 and members with
+ penalties of 0 and 1 are in the queue since the member with
+ penalty 1 will still receive calls. This patch adjusts the
+ special disabled value to be INT_MAX instead of 0. (closes issue
+ ASTERISK-20862) Review: https://reviewboard.asterisk.org/r/2995/
+ Reported by: Schmooze Com
+
+2013-11-08 22:46 +0000 [r402604] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: keep
+ same local (from) tag for outgoing register requests For outbound
+ register requests the tag on the From line was updated every 20
+ seconds prior to a successful registration and also once for each
+ registration renewal. That behavior can possibly cause the
+ registration to be denied because of the different tag, and is
+ not aligned with the intention of RFC 3261 8.1.3.5 "... request
+ constitutes a new transaction and SHOULD have the same value of
+ the Call-ID, To, and From of the previous request...". This
+ updates chan_sip to have a field to keep the local tag in the
+ registration structure and use that tag for registration requests
+ where the callid is also unchanged. (closes issue ASTERISK-12117)
+ Reported by: Pawel Pierscionek Review:
+ https://reviewboard.asterisk.org/r/2988/
+
+2013-11-05 15:08 +0000 [r402468] Kevin Harwell <kharwell at digium.com>
+
+ * channels/chan_sip.c: chan_sip: notify dialog info ignores
+ presentation indicator in callerid The presentation indicator in
+ a callerid (e.g. set by dialplan function
+ Set(CALLERID(name-pres)= ...)) is not checked when SIP Dialog
+ Info Notifies are generated during extension monitoring. Added a
+ check to make sure the name and/or number presentations on the
+ callee (remote identity) are set to allow. If they are restricted
+ then "anonymous" is used instead. (closes issue AST-1175)
+ Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/2976/
+
+2013-10-31 15:57 +0000 [r402287] Matthew Jordan <mjordan at digium.com>
+
+ * main/loader.c: core/loader: Don't call dlclose in a while loop
+ For awhile now, we've noticed continuous integration builds
+ hanging on CentOS 6 64-bit build agents. After resolving a number
+ of problems with symbols, strange locks, and other shenanigans,
+ the problem has persisted. In all cases, gdb shows the Asterisk
+ process stuck in loader.c on one of the infinite while loops that
+ calls dlclose repeatedly until success. The documentation of
+ dlclose states that it returns 0 on success; any other value on
+ error. It does not state that repeatedly calling it will
+ eventually clear those errors. Most likely, the repeated calls to
+ dlclose was to force a close by exhausting the references on the
+ library; however, that will never succeed if: (a) There is some
+ fundamental error at work in the loaded library that precludes
+ unloading it (b) Some other loaded module is referencing a symbol
+ in the currently loaded module This results in Asterisk sitting
+ forever. Since we have matching pairs of dlopen/dlclose, this
+ patch opts to only call dlclose once, and log out as an ERROR if
+ dlclose fails to return success. If nothing else, this might help
+ to determine why on the CentOS 6 64-bit build agent things are
+ not closing successfully. Review:
+ https://reviewboard.asterisk.org/r/2970
+
+2013-10-29 23:41 +0000 [r402224] Rusty Newton <rnewton at digium.com>
+
+ * sounds/Makefile: Updates for 1.4.25 core sounds and 1.4.14 extra
+ sounds, plus new en_GB language set The new sound packages relate
+ to issues: ASTERISK-22544, ASTERISK-22411, ASTERISK-21413,
+ ASTERISK-20782 Modified sounds/Makefile for the new sound
+ versions and to account for the new en_GB language set. (issue
+ ASTERISK-22659) (closes issue ASTERISK-22659) (closes issue
+ ASTERISK-22411) (closes issue ASTERISK-22544)
+
+2013-10-29 14:52 +0000 [r402192] David M. Lee <dlee at digium.com>
+
+ * configure, configure.ac, makeopts.in, Makefile: Backport r373119
+ from 11 to go along with RAII_VAR support. In order to use nested
+ functions on some versions of GCC (e.g. GCC on OS X), the
+ -fnested-functions flag must be passed to the compiler. This
+ patch adds detection logic to ./configure to add the flag if
+ necessary.
+
+2013-10-29 12:40 +0000 [r402150] Matthew Jordan <mjordan at digium.com>
+
+ * main/translate.c, main/xmldoc.c, main/channel.c, main/pbx.c:
+ Remove some spammy debug messages; improve clarity of others
+ Debug messages aren't free. Even when the debug level is
+ sufficiently low such that the messages are never evaluated,
+ there is a cost to having to parse Asterisk logs that contain
+ debug messages that (a) fail to convey sufficient information or
+ (b) occur so frequently as to be next to meaningless. Based on
+ having to stare at lots of DEBUG messages, this patch makes the
+ following changes: * channel.c: When copying variables from a
+ parent channel to a child channel, specify the channels involved.
+ Do not log anything for a variable that is not inherited; the
+ fact that it doesn't have an _ or __ already signifies that it
+ won't be inherited. * pbx.c: Specify what function evaluation has
+ occurred that created the result. * translate.c: Bump up the
+ translator path messages to 10. I've never once had to use these
+ debug messages, and for each format that is registered (on
+ startup) and unregistered (on shutdown) the entire f^2 matrix is
+ logged out. For short tests in the Asterisk Test Suite, this
+ should make finding the actual test much easier. * xmldoc.c: The
+ debug message that 'blah' is not found in the tree is expected.
+ Often, description elements - which are not required - are not
+ provided. This debug message adds no additional value, as it is
+ not indicative of an error or helpful in debugging which element
+ did not contain a 'blah' element as a child. If an element is
+ supposed to contain a child element, then that XML tree should
+ have failed validation in the first place. Review:
+ https://reviewboard.asterisk.org/r/2966/
+
+2013-12-17 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.25.0 Released.
+
+2013-12-16 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.25.0-rc2 Released.
+
+ * AST-2013-006 - app_sms: BufferOverflow when receiving odd length 16
+ bit message
+
+ This patch prevents an infinite loop overwriting memory when a
+ message is received into the unpacksms16() function, where the length
+ of the message is an odd number of bytes.
+ (closes issue ASTERISK-22590)
+
+ * AST-2013-007 - security: Inhibit execution of privilege escalating
+ functions
+
+ This patch allows individual dialplan functions to be marked as
+ 'dangerous', to inhibit their execution from external sources.
+
+ A 'dangerous' function is one which results in a privilege
+ escalation. For example, if one were to read the channel variable
+ SHELL(rm -rf /) Bad Things(TM) could happen; even if the external
+ source has only read permissions.
+
+ Execution from external sources may be enabled by setting
+ 'live_dangerously' to 'yes' in the [options] section of
+ asterisk.conf. Although doing so is not recommended.
+
+ (closes issue ASTERISK-22905)
+
+2013-10-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.25.0-rc1 Released.
+
+2013-10-25 21:51 +0000 [r401959-402000] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
+ rtp payloads copy and improve argument names In function
+ ast_rtp_instance_early _bridge_make_compatible the use of
+ instance 0/1 as arguments doesn't clearly communicate a direction
+ that the copying of payloads from the source channel to the
+ destination channel will occur, making it more probable to have
+ the arguments to ast_rtp_codecs_payloads_copy() put in the
+ reverse order. This patch renames the arguments with _dst and
+ _src suffixes and corrects the copy direction.
+
+ * include/asterisk/pbx.h, main/pbx.c: pbx.c: fix confused match
+ caller id that deleted exten still in hash This fixes a bug where
+ a zero length callerid match adjacent to a no match callerid
+ extension entry would be deleted together, which then resulted in
+ hashtable references to free'd memory. A third state of the
+ matchcid value has been added to indicate match to any extension
+ which allows enforcing comparison of matchcid on/off without
+ errors. (closes issue AST-1235) Reported by: Guenther Kelleter
+ Review: https://reviewboard.asterisk.org/r/2930/
+
+2013-10-25 17:21 +0000 [r401619-401914] Jonathan Rose <jrose at digium.com>
+
+ * utils/clicompat.c: Put clicompat-r2.patch back in We've figured
+ out how to resolve the problems this was causing in 12/trunk, so
+ this can go back in now. (issue ASTERISK-22467) Reported by:
+ Corey Farrell Patches: clicompat-r2.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * utils/clicompat.c: revert clicompat-r2.patch from r401704 Patch
+ caused the following build errors against testsuite
+ https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
+ (issue ASTERISK-22467) Reported by: Corey Farrell
+
+ * main/utils.c: utils: Fix memory leaks and missed unregistration
+ of CLI commands on shutdown Final set of patches in a series of
+ memory leak/cleanup patches by Corey Farrell (closes issue
+ ASTERISK-22467) Reported by: Corey Farrell Patches:
+ main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
+ main-utils-11.patch uploaded by coreyfarrell (license 5909)
+ main-utils-12up.patch uploaded by coreyfarrell (license 5909)
+
+ * tests/test_linkedlists.c: test_linkedlists: Fix memory leak
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ test_linkedlists-1.8.patch uploaded by coreyfarrell (license
+ 5909) test_linkedlists-11up.patch uploaded by coreyfarrell
+ (license 5909)
+
+ * main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
+ reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ jitterbuf-jb_reset-leak-1.8.patch
+ jitterbuf-jb_reset-leak-11up.patch
+
+ * main/astobj2.c: astobj2: Unregister debug CLI commands at exit
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
+ (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * apps/app_voicemail.c: app_voicemail: Memory Leaks against tests
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
+ app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
+
+ * main/asterisk.c, utils/clicompat.c, channels/chan_dahdi.c,
+ codecs/ilbc/doCPLC.c, main/data.c, main/app.c: memory leaks:
+ Memory leak cleanup patch by Corey Farrell (second set) Also
+ covers ast_app_parse_timelen-fail-zero-length.patch, but the
+ patch was replaced with one of my own. (issue ASTERISK-22467)
+ Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch
+ uploaded by coreyfarrell (license 5909) clicompat-r2.patch
+ uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch
+ uploaded by coreyfarrell (license 5909)
+ data-cleanup-test-registration.patch uploaded by coreyfarrell
+ (license 5909) main-asterisk-kill-listener.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * tests/test_dlinklists.c, funcs/func_math.c,
+ channels/sip/reqresp_parser.c, main/test.c,
+ main/editline/readline.c: memory leaks: Memory leak cleanup patch
+ by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
+ Corey Farrell Patches:
+ chan_sip-parse_contact_header_test-free-contacts.patch uploaded
+ by coreyfarrell (license 5909) cli-filename-completion-leak.patch
+ uploaded by coreyfarrell (license 5909) func_math.patch uploaded
+ by corefarrell (license 5909) main-test-cleanup.patch uploaded by
+ coreyfarrell (license 5909) test_dlinklists.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
+ Address jittery DTMF events in RTP streams (closes issue
+ ASTERISK-21170) Reported by: NITESH BANSAL Patches:
+ dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
+ Review: https://reviewboard.asterisk.org/r/2938/
+
+2013-10-23 16:34 +0000 [r401577] Richard Mudgett <rmudgett at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a filter
+ when the CDR value is empty. Extra CDR records are written if a
+ filtered CDR value is empty because the filter is not checked.
+ (closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria
+
+2013-10-23 15:19 +0000 [r401537] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_mgcp.c: chan_mgcp: Properly handle malformed media
+ lines This corrects a situation in which a media line was not
+ parsed properly and resulted in a crash. (closes issue
+ ASTERISK-21190) Reported by: adomjan Patches:
+ chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
+
+2013-10-23 11:10 +0000 [r401497] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix an issue where an incompatible audio
+ format may be added to SDP. If preferred codecs included any
+ non-audio format the code would mistakenly add the audio format,
+ even if it was not a joint capability with the remote side.
+ (closes issue ASTERISK-21131) Reported by: nbougues Patches:
+ patch_unsupported_codec_1.8.patch uploaded by nbougues (license
+ 6470)
+
+2013-10-22 22:36 +0000 [r401445] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP is
+ not available during SSRC change In r400089, a patch was put in
+ to correct erroneous RTCP statistic resets. Unfortunately,
+ ast_rtp_read can be called on an RTP instance that does not have
+ RTCP information. This patch prevents that crash by only
+ resetting the statistics if we do actually have an RTCP instance.
+ (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
+ Bigelow
+
+2013-10-22 00:13 +0000 [r401378] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_analog.c: chan_dahdi: Fix unable to get index
+ warning when transferring an analog call. Transferring an analog
+ call using flashhooks generated an unable to get index WARNING
+ message when the transfer is completed. * Removed unnecessary
+ analog subchannel shell games when transferring a call using
+ flashhooks. Thanks to Tzafrir Cohen for mentioning this in a
+ comment on issue ASTERISK-22720.
+
+2013-10-21 19:45 +0000 [r401325] Kevin Harwell <kharwell at digium.com>
+
+ * main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
+ tgetstr(), when libncurses5-dev isn't installed Include the
+ appropriate declarations when not using termcap, but term+curses
+ and [n]curses do not exist. (closes issue ASTERISK-22351)
+ Reported by: A. Iglesias Patches:
+ issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
+ by wdoekes (license 5674)
+
+2013-10-18 14:40 +0000 [r401178] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/channel.c: Properly copy/remove the device state cache flag
+ over a masquerade. In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE
+ flag was added that tells the devstate system to not cache states
+ for non-real devices. However, when optimizing away channels
+ (ast_do_masquerade), that flag wasn't copied. In my case, using
+ Local devices as queue members created a situation where the
+ endpoint was considered in use, but the state change of the
+ device being available again was ignored (not cached). The
+ endpoint channel was optimized into the (previously) Local
+ channel, but kept the do-not-cache flag. The end result being
+ that the queue member apparently stayed in use forever. (closes
+ issue ASTERISK-22718) Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/2925/
+
+2013-10-17 15:22 +0000 [r401119] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_jabber.c: Reduce log level of a non-pubsub error message
+ Drop an error log message to debug level 1 since distributed
+ device state functions correctly when receiving this message and
+ it spams the logs. (closes issue ASTERISK-22410) Reported by:
+ abelbeck Patches:
+ asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
+ uploaded by abelbeck (License 5903)
+
+2013-10-16 11:04 +0000 [r401049] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_queue.c: Don't check all realtime queues when doing
+ "queue show some_queue". When using realtime queues, queues have
+ to be fetched from the database every now and then to see if any
+ info has been changed or to see if the queue has been removed.
+ When fetching info for an individual queue, the pruning of other
+ queues is unnecessarily costly. Review:
+ https://reviewboard.asterisk.org/r/2907/
+
+2013-10-15 14:52 +0000 [r400970] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Prevent chan_sip from sending duplicate
+ BYEs. When a 200 OK for an initial INVITE is received, we were
+ doing the right thing by ACKing and sending an immediate BYE.
+ However, we also were doing the wrong thing and queuing an answer
+ frame, thus causing the call to be answered. This would cause the
+ call to be hung up by the channel thread, thus resulting in a
+ second BYE being sent out. In this fix, I also have set the
+ hangupcause to be correct since the initial BYE being sent by
+ Asterisk had an unknown hangup cause. I have changed to using
+ "Bearer capabilty not available" since the call was hung up due
+ to an SDP offer/answer error. (closes issue ASTERISK-22621)
+ reported by Kinsey Moore
+
+2013-10-14 21:40 +0000 [r400907] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: Reflect the set software gain
+ in the CLI "dahdi show channel" output. * Remember the swgain
+ setting from CLI "dahdi set swgain" command so the CLI "dahdi
+ show channel" output will reflect the current setting. * Updated
+ CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
+ (issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
+ jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded
+ by rmudgett
+
+2013-10-14 21:32 +0000 [r400906] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Do not increment the SDP version between 183
+ and 200 responses. Bumping the SDP version number can cause
+ interoperability problems since receivers of the responses will
+ expect that a 200 SDP will be identical to a previous 183 SDP.
+ (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches:
+ dont-increment-session-version-in-2xx-after-183.patch uploaded by
+ NITESH BANSAL (License #6418)
+
+2013-10-08 22:26 +0000 [r400694-400767] Kinsey Moore <kmoore at digium.com>
+
+ * configure, configure.ac: Add warning when compiling with iODBC
+ support When running configure, libiodbc2 development headers
+ will fulfill the requirement for ODBC development headers, but
+ will not function properly. This adds a warning when libiodbc2
+ development headers are detected instead of unixodbc development
+ headers. (closes issue ASTERISK-22459) Reported by: Patrick
+ Maille Tested by: Walter Doekes Patches:
+ issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
+ (License 5674)
+
+ * funcs/func_config.c: Fix func_config list entry allocation The
+ AST_CONFIG dialplan function defined in func_config.c allocates
+ its config file list entries using ast_malloc. List entry
+ allocations destined for use with Asterisk's linked list API must
+ be ast_calloc()d or otherwise initialized so that list pointers
+ are set to NULL. These uses of ast_malloc have been replaced by
+ ast_calloc to prevent dereferencing of uninitialized pointer
+ values when traversing the list. (closes issue ASTERISK-22483)
+ Reported by: Brian Scott
+
+2013-10-06 17:07 +0000 [r400622] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_queue.c: Fix Regression With Queuelog EXITWITHKEY Only
+ Logging Two Out Of Four Fields Commit r62462 added two extra
+ fields for logging "the original position the caller entered the
+ queue at, and the amount of time the caller was waiting in the
+ queue." But when r75969 was merged from 1.4 into trunk (r75977),
+ these two fields disappeared. Those two extra fields were not
+ logged in 1.4 and when the patch was merged, those fields went
+ away. Therefore, this is a regression and was caught by the
+ reporter because he was reading the awesome "Asterisk: The
+ Definitive Guide" book. (closes issue ASTERISK-22197) Reported
+ by: Dalius M. Tested by: Dalius M. Patches:
+ asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2901/
+
+2013-10-03 22:51 +0000 [r400469] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Don't ignore expires value in
+ contact header if it lacks semicolon (closes issue
+ ASTERISK-22574) Reported by: Filip Jenicek Patches:
+ chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
+
+2013-10-03 18:25 +0000 [r400393] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_multicast.c: Ensure res_rtp_mutlicast sets SSRC
+ properly This fixes a bug where the SSRC field on multicast RTP
+ can be stuck at 0 which can cause problems for endpoints trying
+ to make sense of incoming streams. (closes issue ASTERISK-22567)
+ Reported by: Simone Camporeale Patches:
+ 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
+ (License 6536)
+
+2013-10-02 21:30 +0000 [r400314] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_iax2.c: Cast Integer Argument To Unsigned Char The
+ member reg in the peercnt structure is an unsigned char and
+ peercnt_modify() is expecting an unsigned char argument which
+ gets assigned to peercnt->reg. This patch fixes that by casting
+ the integer argument being passed to peercnt_modify to unsigned
+ char.
[... 46476 lines stripped ...]
More information about the svn-commits
mailing list