[asterisk-commits] bebuild: tag 1.8.27.0-rc1 r411549 - /tags/1.8.27.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 28 13:24:36 CDT 2014
Author: bebuild
Date: Fri Mar 28 13:24:31 2014
New Revision: 411549
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411549
Log:
Importing files for 1.8.27.0-rc1 release.
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tags/1.8.27.0-rc1/ChangeLog (with props)
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--- tags/1.8.27.0-rc1/ChangeLog (added)
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+2014-03-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.27.0-rc1 Released.
+
+2014-03-28 16:16 +0000 [r411462] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/http.c, main/tcptls.c, main/manager.c: http: response body
+ often missing after specific request This patch works around a
+ problem with the HTTP body being dropped from the response to a
+ specific client and under specific circumstances: a) Client
+ request comes from node.js user agent "Shred" via use of
+ swagger-client library. b) Asterisk and Client are *not* on the
+ same host or TCP/IP stack In testing this problem, it has been
+ determined that the write of the HTTP body is lost, even if the
+ data is written using low level write function. The only solution
+ found is to instruct the TCP stack with the shutdown function to
+ flush the last write and finish the transmission. See review for
+ more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
+ Reported by: Sam Galarneau Review:
+ https://reviewboard.asterisk.org/r/3402/
+
+2014-03-28 15:42 +0000 [r411372-411457] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between 1.4
+ and 1.8+ systems.
+
+ * res/res_config_odbc.c, res/res_odbc.exports.in, UPGRADE.txt,
+ res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h: res_config_odbc/res_odbc: Fix
+ handling of non-text columns updates with empty values. This
+ patch fixes setting nullable integer columns to NULL instead of
+ an empty string, which fails for PostgreSQL, for example. The
+ current code is supposed to do so, but the check is broken. The
+ patch also allows the first column in the list to be a nullable
+ integer. This patch also adds a compatibility setting in
+ res_odbc.conf, allow_empty_string_in_nontext. It is enabled by
+ default. It should be disabled for database backends (such as
+ PostgreSQL) that require NULL instead of an empty string for
+ Integer columns. Review: https://reviewboard.asterisk.org/r/3375
+ (issue ASTERISK-23459) Reported by: zvision patches:
+ res_config_odbc.diff uploaded by zvision (License 5755)
+
+ * channels/sip/include/sip.h: chan_sip: Add MESSAGE request to
+ allowed methods The allowed methods advertised by chan_sip did
+ not previously note the MESSAGE request. Even in Asterisk 1.8, we
+ do accept in-dialog MESSAGE requests; we should advertise that we
+ support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
+ #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
+ Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
+ Review: https://reviewboard.asterisk.org/r/3396/
+
+2014-03-27 19:06 +0000 [r411313] Corey Farrell <git at cfware.com>
+
+ * funcs/func_groupcount.c, funcs/func_callcompletion.c,
+ funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
+ funcs/func_frame_trace.c, funcs/func_channel.c,
+ funcs/func_blacklist.c, funcs/func_callerid.c, apps/app_stack.c,
+ res/res_calendar.c, apps/app_jack.c, funcs/func_speex.c,
+ funcs/func_dialplan.c, channels/chan_sip.c, funcs/func_math.c,
+ apps/app_readexten.c, funcs/func_strings.c, res/res_jabber.c,
+ channels/chan_iax2.c, res/res_mutestream.c, funcs/func_global.c,
+ apps/app_speech_utils.c: Fix dialplan function NULL channel
+ safety issues (closes issue ASTERISK-23391) Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/3386/
+
+2014-03-26 22:43 +0000 [r411243] Joshua Colp <jcolp at digium.com>
+
+ * main/say.c: say: Fix a bug where SayNumber in Polish tries to
+ play incorrect sound. This change fixes a bug where calling
+ SayNumber with a number divisible by 100 using the Polish
+ language would cause the code to attempt to play a sound file
+ with an empty name. (closes issue ASTERISK-23509) Reported by:
+ zvision Review: https://reviewboard.asterisk.org/r/3378/
+
+2014-03-26 15:50 +0000 [r411189] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send real
+ CallerID information with P-Assserted-Identity (RFC-3325) Prior
+ too this patch, the P-Asserted-Identity header would include
+ anonymous caller id information which seems to go against the
+ point of the P-Asserted-Identity header. Now the real caller ID
+ information will be included in this header. Also, no privacy
+ header would be included. This patch adds 'Privacy: id' to
+ outgoing SIP messages that include the P-Asserted-Identity
+ header. (closes issue AST-1301)
+
+2014-03-25 15:50 +0000 [r411088] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
+ update_provisional_keepalive() is called while
+ send_provisional_keepalive_full() is waiting on the PVT lock,
+ then pvt->provisional_keepalive_sched_id will be changed to a new
+ sched_id value by update_provisional_keepalive(), but that new
+ sched_id then may be overwritten with -1 by
+ send_provisional_keepalive_full(), killing the pvt's reference to
+ a schedule and "leaking" the reference. (closes issue
+ ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
+ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
+ Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
+ (license 5012)
+
+2014-03-24 21:36 +0000 [r411021] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Always use fromdomain if set for
+ domain, even if callerid is set to restricted. (closes issue
+ ASTERISK-20841) Reported by: Kelly Goedert
+
+2014-03-17 21:54 +0000 [r410710-410748] Russ Meyerriecks <rmeyerreicks at digium.com>
+
+ * main/callerid.c: !fixup: callerid: Logic error in checksum
+ processing Fixes syntax error in previous commit :-(
+
+ * main/callerid.c: callerid: Logic error in checksum processing
+ Callerid checksum-ing was being handled incorrectly here. When
+ the checksum is calculated to be 0x00, it will perform 0x100-0x00
+ which results in 0x100. This value will then fail the otherwise
+ correct callerid message. This patch changes the logic to simply
+ add the calculated checksum to the transmitted 2's compliment
+ checksum. Review: https://reviewboard.asterisk.org/r/3356/
+ (closes issue ASTERISK-23488)
+
+2014-03-10 17:00 +0000 [r410380] Richard Mudgett <rmudgett at digium.com>
+
+ * main/http.c: AST-2014-001: Stack overflow in HTTP processing of
+ Cookie headers. Sending a HTTP request that is handled by
+ Asterisk with a large number of Cookie headers could overflow the
+ stack. Another vulnerability along similar lines is any HTTP
+ request with a ridiculous number of headers in the request could
+ exhaust system memory. (closes issue ASTERISK-23340) Reported by:
+ Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
+ Manuel Sadosky, Buenos Aires, Argentina
+
+2014-03-10 13:15 +0000 [r410308] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
+ session timers request This change allows chan_sip to avoid
+ creation of the channel and consumption of associated file
+ descriptors altogether if the inbound request is going to be
+ rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
+ Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
+ Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
+ Corey Farrell (license 5909)
+
+2014-03-07 22:50 +0000 [r410224] Corey Farrell <git at cfware.com>
+
+ * channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
+ unload_module and do_monitor Release monlock before calling
+ pthread_join. This ensures do_monitor cannot freeze by locking
+ monlock during module unload. (closes issue ASTERISK-21406)
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/3284/
+
+2014-03-07 04:35 +0000 [r410105] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Allow static realtime members to
+ be qualified during module load. When a static realtime peer with
+ qualify=yes is loaded, Asterisk will fail to send an OPTIONS
+ request due to the lastms being equal to 0. This results in the
+ peer being unable to receive calls from Asterisk because the
+ status is permanently UNKNOWN. This patch allows an OPTIONS
+ request to be sent during module load by ignoring the lastms
+ value on startup only. Review:
+ https://reviewboard.asterisk.org/r/3294/ (closes issue
+ ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
+ wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
+ Peirce (license 6112)
+
+2014-03-06 23:01 +0000 [r410043] Russell Bryant <russell at russellbryant.com>
+
+ * res/res_musiconhold.c: moh: fix a refcount error with realtime
+ MOH I observed a crash in res_musiconhold on an Asterisk 11
+ system using realtime MOH. Investigation of the backtrace showed
+ a corrupt mohclass, implying that it got destroyed before the
+ code expected it to. I went looking for reference counting errors
+ that could have caused this crash and this patch this result. It
+ contains 2 changes. 1) Remove a usless block of code that was
+ impossible to reach. There was even a comment indicating that it
+ was impossible to reach. The conditional includes
+ "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
+ inside of an if block with the opposite check
+ "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
+ good reason to keep it around. 2) A similar block to #1 contained
+ a reference counting error. It stores state->class in the local
+ variable mohclass without increasing its reference count. The
+ reference count on mohclass is decremented at the end of the
+ function. This block of code probably very rarely runs, which
+ would help explain why this system was working fine for many
+ months before experiencing a crash. Review:
+ https://reviewboard.asterisk.org/r/3282/
+
+2014-03-05 20:31 +0000 [r409916] Kinsey Moore <kmoore at digium.com>
+
+ * main/config.c: config: Fix inverted test The test of the result
+ of the stat() call was inverted such that its output was only
+ used if the call failed. This inverts the test so that the output
+ of stat() is used correctly. This was causing full reloads on
+ unchanged files. (closes issue ASTERISK-23383) Reported by: David
+ Woolley
+
+2014-03-05 16:50 +0000 [r409833] David M. Lee <dlee at digium.com>
+
+ * main/config.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Corrected cross-platform stat nanosecond code When
+ nanosecond time resolution was added for identifying config file
+ changes, it didn't cover all of the myriad of ways that one might
+ obtain nanosecond time resolution off of struct stat. Rather than
+ complicate the #if even further figuring out one system from the
+ next, this patch directly tests for the three struct members I
+ know about today, and #ifdef's accordingly. Review:
+ https://reviewboard.asterisk.org/r/3273/
+
+2014-03-05 12:04 +0000 [r409777] Sean Bright <sean at malleable.com>
+
+ * contrib/scripts/astgenkey, contrib/scripts/astgenkey.8: Fix
+ references to 'keys' CLI commands in astgenkey
+
+2014-03-05 05:10 +0000 [r409705] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Add update_peer function to
+ unistim_rtp_glue, improve other unistim_rtp_glue functions
+ conforming to other channel drivers. Do not forget auto-detected
+ and user-selected phone settings on 'unistim reload'
+
+2014-03-04 19:32 +0000 [r409623] Michael L. Young <elgueromexicano at gmail.com>
+
+ * funcs/func_audiohookinherit.c: func_audiohookinheritance: Check
+ If A Channel Was Specified This patch prevents a crash when using
+ the function audiohookinheritance without setting the channel.
+ (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by:
+ Joel Vandal Patches:
+ asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/3272/
+
+2014-03-04 16:50 +0000 [r409521-409566] Kinsey Moore <kmoore at digium.com>
+
+ * main/astobj2.c: AO2: Add an assert for bad objects This adds an
+ assert that will only be active if Asterisk is compiled with
+ DO_CRASH and allows the testsuite to fail tests that would
+ otherwise require log file parsing.
+
+ * main/rtp_engine.c: rtp_engine: Clean up after a failed remote
+ bridge Upon failure of an INVITE transaction meant to initiate a
+ remote native bridge, rtp_engine.c would not clean up
+ non-reference-counted bridge instance pointers leaving a dangling
+ pointer which was being used to perform a local native bridge
+ after the other channel had hung up. This lead to dereferencing
+ into freed memory and plenty of AO2 errors. This change allows
+ the remote native bridge loop to clean up properly when the
+ bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy
+ Laine
+
+2014-03-04 14:50 +0000 [r409472] Sean Bright <sean at malleable.com>
+
+ * channels/chan_sip.c: Minor whitespace change to 'sip show peers'
+ output. (closes issue ASTERISK-23406) Reported by: ibercom Tested
+ by: ibercom Patches: asterisk-11.patch uploaded by ibercom
+
+2014-03-04 13:39 +0000 [r409436] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * Makefile: buildsystem: Unbreak 'make -qp' on 1.8. r408083 caused
+ trouble with make -qp. Backport r408193 to 1.8 as well. (closes
+ issue ASTERISK-23382) Reported by: Corey Farrell
+
+2014-03-03 02:06 +0000 [r409361] Matthew Jordan <mjordan at digium.com>
+
+ * main/asterisk.c: doxygen: Tweak the link back to ye olde Digium
+ website
+
+2014-03-02 10:58 +0000 [r409308] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a legal
+ option of gcc. Unofficially gcc considers it to be equivalent of
+ -O3. clang chalks on it, though. This commit sets the default
+ optimization flag to be -O3, like gcc actually considered it.
+ Review: https://reviewboard.asterisk.org/r/3280/
+
+2014-02-28 21:00 +0000 [r409156-409207] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Add precautionary p->owner checks.
+ * Add precautionary p->owner checks in sip_hangup(),
+ get_refer_info(), get_also_info(), and
+ interpret_t38_parameters(). * Simplify some tangled logic in
+ get_refer_info(), get_also_info(), and add_rpid(). * Removed some
+ dead code in handle_request_invite(). (closes issue
+ ASTERISK-23323) Reported by: Walter Doekes Patches:
+ issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-11.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-12.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-trunk.patch (license #5674)
+ uploaded by wdoekes (modified)
+
+ * channels/chan_sip.c: chan_sip: Fix crash in
+ ast_channel_hangupcause_set(). * Fix crash in
+ ast_channel_hangupcause_set() because p->owner not checked before
+ calling. Regression introduced by the fix for ASTERISK-22621.
+ (closes issue ASTERISK-23135) Reported by: OK (issue
+ ASTERISK-23323) Reported by: Walter Doekes
+
+2014-02-27 16:23 +0000 [r409077] David M. Lee <dlee at digium.com>
+
+ * utils/astman.c: Fix memory stomping bug in astman. This memset
+ complained in dev mod on my Ubuntu box. The memset is both
+ unnecessary and dangerous. At this point, m hasn't been
+ initialized yet, so the memset will write off to whatever address
+ happens to be on the stack at the time.
+
+2014-02-27 15:59 +0000 [r409052] Corey Farrell <git at cfware.com>
+
+ * res/res_fax.c, configs/res_fax.conf.sample: res_fax: Warn that
+ minrate=2400 is not valid for V.27 instead of failing load.
+ Change minrate from 2400 to 4800 on config reload in response to
+ changes from ASTERISK-22790 only. Any config with minrate of 2400
+ that would fail before r405693 will still fail. Comment out many
+ settings in res_fax.conf.sample. The defaults are set in
+ res_fax.c, so setting the same value in sample config does
+ nothing but make the sample config more fragile. (closes issue
+ ASTERISK-23231) Reported by: David Brillert Review:
+ https://reviewboard.asterisk.org/r/3261/
+
+2014-02-27 12:39 +0000 [r409001] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
+ crash during remote native bridging when calling get_codecs When
+ two RTP channels are in a remote bridge, the remote bridging loop
+ in rtp_engine will periodically check to see if the two channels
+ can still be bridged. One of the many things it checks is whether
+ or not the codecs have changed on the channel. If the codec has
+ changed, it will break out of the loop to re-determine which type
+ of bridge is appropriate. In order to perform this check, the
+ ast_rtp_glue virtual table's get_codec callback is called for
+ each channel. The callback implementations assume that the
+ channel tech private is valid when called; as such, there has
+ always been some code in place to check whether or not the
+ channel pvt is NULL before calling. However, this check is
+ insufficient. The channels are unlocked during the remote
+ bridging loop. It is possible for a channel to get masqueraded
+ between the check for the pvt being NULL and the actual call to
+ get_codec. When this occurs, the callback is called with a ZOMBIE
+ channel, which now has a NULL pvt. Crash. While this has always
+ been possible in Asterisk 1.8, it is much more likely to occur in
+ Asterisk 11 and later versions due to the timing changes that
+ occur when getting the codec from a channel. Note that this is
+ much more likely to be reproduced on slow, boggy hardware running
+ Asterisk 11 - but fairly rarely otherwise. Also Note: This crash
+ was also caught by the various SIP blind transfer tests, in
+ addition to the bug report Alec filed. Review:
+ https://reviewboard.asterisk.org/r/3247/ (closes issue
+ ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis
+
+2014-02-25 17:41 +0000 [r408876] Rusty Newton <rnewton at digium.com>
+
+ * configs/voicemail.conf.sample: configs/voicemail.conf.sample -
+ Make mailcmd sample text more explicit Made the wording a bit
+ more explicit. Didn't really change the meaning.
+
+2014-02-22 02:26 +0000 [r408785] Corey Farrell <git at cfware.com>
+
+ * utils/extconf.c, utils/conf2ael.c, res/ael/pval.c, main/pbx.c:
+ Remove extra defines of AST_PBX_MAX_STACK. * Ensure
+ AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
+ incorrect function parameters in utils/extconf.c. (closes issue
+ ASTERISK-23141) Reported by: Maxim Review:
+ https://reviewboard.asterisk.org/r/3241/
+
+2014-02-21 20:18 +0000 [r408642-408747] Kevin Harwell <kharwell at digium.com>
+
+ * apps/app_forkcdr.c: app_forkcdr: ForkCDR v option does not keep
+ CDR variables for subsequent records When the 'v' option is
+ specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is
+ set only for the first CDR in the chain. So ForkCDR works fine
+ with this option only once. After the second and further calls to
+ ForkCDR, CDR variables get cleared on all CDRs besides the first
+ one and moved to the newly forked CDR. It always sets the
+ KEEP_VARS flag on the first CDR in the chain, instead of the most
+ recent CDR which is used as a base to fork a new CDR. This patch
+ sets KEEP_VARS flag on the most recent CDR on the stack (the CDR
+ used for forking). (closes issue ASTERISK-23260) Reported by:
+ zvision Patches: app_forkcdr.diff uploaded by zvision (license
+ 5755)
+
+ * main/rtp_engine.c: rtp_engine: Output mixup in
+ ${CHANNEL(rtpqos,audio,all)} Fixed the output of
+ CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
+ (closes issue ASTERISK-23261) Reported by: rsw686 Patches:
+ rtpqos.patch uploaded by rsw686 (license 5887)
+
+ * channels/chan_sip.c, main/channel.c: channel.c: MOH is not
+ working for transferee after attended transfer Updated the code
+ to check to see if MOH is playing on the transferor and if so
+ then start it on the channel that replaces it during a
+ masquerade. Example scenario of the problem: Alice calls Bob and
+ then Bob begins the attended transfer process into a queue. Upon
+ going on hold Alice hears music and so does Bob once he is in the
+ queue. Bob then transfers Alice into the queue and then music for
+ Alice stops even though she should be hearing it since has now
+ replaced Bob in the queue. The problem that was occurring is that
+ once the channel was masqueraded the app (queues, confbridge,
+ etc...) had no way of knowing that the channel had just been
+ swapped out thus it did not start music for the present channel.
+ Credit to Olle Johansson for pointing me in the right direction
+ on this issue. (closes issue ASTERISK-19499) Reported by: Timo
+ Teräs Review: https://reviewboard.asterisk.org/r/3226/
+
+2014-02-21 10:35 +0000 [r408589] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
+ variables
+
+2014-02-21 00:46 +0000 [r408536] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_chanspy.c: app_chanspy: Documentation Update To Clarify
+ "x" Option When using the "x" option (specify a DTMF digit to
+ exit the application), it is not obvious in the documentation
+ that this only works when spying on a channel. If a channel being
+ used to spy on other channels is waiting to connect to a channel
+ or is no longer attached to a channel, the DTMF is ignored. As
+ noted on the issue tracker, since there are workarounds available
+ and this is a rarely used option we are opting for a
+ documentation change here. (closes issue ASTERISK-22661) Reported
+ by: Chris Hillman Patches:
+ asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2990/
+
+2014-02-20 02:39 +0000 [r408447] Rusty Newton <rnewton at digium.com>
+
+ * apps/app_queue.c: apps/app_queue - Fix incorrect Macro parameter
+ documentation Macro is executed on the called channel, not the
+ calling channel. (closes issue ASTERISK-23069) Reported By: Bryan
+ Anderson
+
+2014-02-19 19:01 +0000 [r408387] Richard Mudgett <rmudgett at digium.com>
+
+ * main/config.c: config: Add file size and nanosecond resolution
+ fields to the cached modified config file information. Repeatedly
+ modifying config files and reloading too fast sometimes fails to
+ reload the configuration because the cached modification
+ timestamp has one second resolution. * Added file size and
+ nanosecond resolution fields to the cached config file
+ modification timestamp information. Now if the file size changes
+ or the file system supports nanosecond resolution the modified
+ file has a better chance of being detected for reload. * Added a
+ missing unlock in an off-nominal code path. (closes issue
+ AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
+
+2014-02-19 11:30 +0000 [r408328] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooCapability.c, addons/ooh323c/src/ooh245.c:
+ process receiveAndTransmit user input remote caps instead of
+ receive only send receiveAndTransmit user input our caps instead
+ of receive only
+
+2014-02-16 03:14 +0000 [r408200] Matthew Jordan <mjordan at digium.com>
+
+ * main/pbx.c: pbx: Handle a completely empty dialplan during a
+ context merge It is highly unlikely, but - at least in Asterisk
+ 12 - theoretically possible to load Asterisk with no dialplan
+ whatsoever. If that occurs, and some other module (that is not a
+ pbx module) attempts to merge its contexts into the dialplan, the
+ existing merge routine will crash. This is because it is not
+ insane, and rightly believes that you provided some sort of
+ dialplan, somewhere. This patch will gracefully merge the
+ contexts in such a case. Note that this is highly unlikely to
+ occur in 1.8/11, as features will most likely provide some
+ dialplan via parking. However, in Asterisk 12, parking is now
+ provided by res_parking, and hence may create its dialplan later.
+ (closes issue ASTERISK-23297) Reported by: CJ Oster Review:
+ https://reviewboard.asterisk.org/r/3222
+
+2014-02-14 21:52 +0000 [r408142] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/pbx.c: pbx: ast_custom_function_unregister resource leak In
+ pbx.c ast_custom_function_unregister(), a list of escalations
+ being removed from the list wasn't being free'd creating a leak.
+ This patch corrects that by freeing the records. Review:
+ https://reviewboard.asterisk.org/r/3213/ Reported by: Corey
+ Farrell Patches: acf_escalating_leak.patch uploaded by
+ coreyfarrell (license 5909)
+
+2014-02-14 13:25 +0000 [r408083] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * Makefile: buildsystem: Don't force main to depend on everything
+ else. Directory 'main' only needs to depend on embedded modules.
+ If no module embedding is selected, the dependency is dropped.
+ Review: https://reviewboard.asterisk.org/r/3212/
+
+2014-02-14 01:22 +0000 [r408020] Rusty Newton <rnewton at digium.com>
+
+ * configs/agents.conf.sample: configs/agents.conf.sample - Remove
+ example for non-functional "goodbye" parameter The "goodbye"
+ parameter is not implemented in the source code, it does nothing.
+ (closes issue SWP-6518) Reported By: Steve Pitts
+
+2014-02-10 16:33 +0000 [r407873] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * res/res_config_pgsql.c: res_config_pgsql: Fix
+ ast_update2_realtime calls. Fix so multiple updates from a single
+ call works (add missing ','). Remove bogus ast_free's that
+ weren't supposed to be there. Moved a few spaces for readability.
+ Review: https://reviewboard.asterisk.org/r/3194/
+
+2014-02-09 15:34 +0000 [r407817] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * channels/chan_dahdi.c, /: chan_dahdi: handle DAHDI_EVENT_REMOVED
+ on a pri D-Channel When a DAHDI device is removed at run-time it
+ sends the event DAHDI_EVENT_REMOVED on each channel. This is
+ intended to signal the userspace program to close the respective
+ file handle, as the driver of the device will need all of them
+ closed to properly clean-up. This event has long since been
+ handled in chan_dahdi (chan_zap at the time). However the event
+ that is sent on a D-Channel of a "PRI" (ISDN) span simply gets
+ ignored. This commit adds handling for closing the file
+ descriptor (and shutting down the span, while we're at it). It
+ also adds a CLI command 'pri destroy span <N>' to destroy the
+ span and its DAHDI channels. Backported from trunk/12. Review:
+ https://reviewboard.asterisk.org/r/726/ ........ Merged revisions
+ 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk
+
+2014-02-07 20:42 +0000 [r407678-407764] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Add some more iaxs[] NULL checks
+ to a routine already full of them.
+
+ * channels/chan_iax2.c, include/asterisk/frame.h,
+ configs/iax.conf.sample: chan_iax2: Block unnecessary control
+ frames to/from the wire. Establishing an IAX2 call between
+ Asterisk v1.4 and v1.8 (or later) results in an unexpected call
+ disconnect. The problem happens because newer values in the enum
+ ast_control_frame_type are not consistent between the branch
+ versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later)
+ using IAX2 2) v1.8 answers and sends a connected line update
+ control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4
+ receives the control frame as an end-of-q (on v1.4
+ AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the
+ receive queue becomes empty. Several things are done by this
+ patch to fix the problem and attempt to prevent it from happening
+ again in the future: * Added a warning at the definition of enum
+ ast_control_frame_type about how to add new control frame values.
+ * Made block sending and receiving control frames that have no
+ reason to go over the wire. * Extended the connectedline iax.conf
+ parameter to also include the redirecting information updates. *
+ Updated the connectedline iax.conf parameter documentation to
+ include a notice that the parameter must be "no" when the peer is
+ an Asterisk v1.4 instance. (closes issue AST-1302) Review:
+ https://reviewboard.asterisk.org/r/3174/
+
+2014-02-07 12:59 +0000 [r407622] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * configs/indications.conf.sample: indications.conf: add stutter
+ tone; end properly * If the "stutter" (voicemail indication) tone
+ is indeed a stutter tone, and it ends with a constant tone, make
+ sure that it is the dial tone. This was done for India (in),
+ Mexico (mx) and the Philippines (ph). * If no "stutter" tone
+ exists for a country, provide one. This was done for Spain (es),
+ Malaysia (my) and Venezuela (ve). Review:
+ https://reviewboard.asterisk.org/r/3158/
+
+2014-02-05 22:58 +0000 [r407511] Rusty Newton <rnewton at digium.com>
+
+ * formats/format_wav.c: formats/format_wav: enhancing log message
+ "Not a wav file" to be clear on what is supported Modifying the
+ log message to be more specific as to what is supported.
+ Specifically it seems format_wav supports only PCM encoded
+ versions with a lower-case '.wav' extension. (closes issues
+ ASTERISK-22310) Reported by: Jim Credland Review:
+ https://reviewboard.asterisk.org/r/3188/
+
+2014-02-05 20:30 +0000 [r407455] Kinsey Moore <kmoore at digium.com>
+
+ * main/logger.c: Logger: Fix handling of absolute paths This fixes
+ path handling for log files so that an extra / is not appended to
+ the file path when the path is absolute (begins with /). This
+ would previously result in different but functionally equivalent
+ paths in the output of 'logger show channels'.
+
+2014-02-04 19:48 +0000 [r407272-407337] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/devicestate.h, main/devicestate.c: devicestate:
+ Make ast_devstate_changed_literal() return value and doxygen
+ consistent. Nothing actually cares about the value anyway.
+ (closes issue ASTERISK-23178) Reported by: Jonathan Rose
+
+ * configs/sip.conf.sample, main/tcptls.c: tcptls.c: Made TLS handle
+ a certificate chain file. Thanks to Guillaume Martres for doing
+ the necessary research to validate the change. (closes issue
+ ASTERISK-17727) Reported by: LN Patches:
+ use_certificate_chain.patch (license #5864) patch uploaded by st
+ documente_certificate_chain.patch (license #6576) patch uploaded
+ by Guillaume Martres
+
+2014-02-04 02:19 +0000 [r407205] Joshua Colp <jcolp at digium.com>
+
+ * res/res_clialiases.c: res_clialiases: Fix crash when reloading
+ and re-aliasing an alias that is in use. The code assumed that
+ unregistering the alias would always succeed while in practice
+ this is not actually true. A common case is the "reload" command
+ itself. If the cli_aliases.conf configuration file was changed
+ and reload executed the command would fail to unregister and
+ ultimately point to freed memory. The reload process now checks
+ whether unregistering succeeded or not and if not the old CLI
+ alias is retained. (closes issue ASTERISK-19773) Reported by:
+ Joel Vandal (closes issue ASTERISK-22757) Reported by: Gareth
+ Blades
+
+2014-02-01 00:22 +0000 [r407100] Corey Farrell <git at cfware.com>
+
+ * apps/app_stack.c: app_stack: protect against missing parameters
+ to STACK_PEEK and LOCAL_PEEK STACK_PEEK requires 2 parameters and
+ LOCAL_PEEK requires 1 parameter. This protects against situations
+ where those parameters are blank or missing by logging an error
+ and returning. (closes issue ASTERISK-23220) Reported by: James
+ Sharp
+
+2014-01-31 23:18 +0000 [r407041] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_dial.c: app_dial: Allow macro/gosub pre-bridge execution
+ to occur on priorities The parsing for the destination of the
+ macro/gosub uses the '^' character to separate out context,
+ extension, and priority. However, the logic for the macro/gosub
+ execution was written such that it would only do the actual
+ macro/gosub jump if a '^' character existed. This doesn't apply
+ when the macro/gosub jump occurs in a priority/priority label.
+ This patch changes the logic so that the parsing still occurs,
+ but the jump will occur even for priorities/priority labels.
+ (issue ASTERISK-23164) Review:
+ https://reviewboard.asterisk.org/r/3154
+
+2014-01-30 20:26 +0000 [r406933] Corey Farrell <git at cfware.com>
+
+ * main/udptl.c, res/res_rtp_asterisk.c: res_rtp_asterisk & udptl:
+ fix port selection to work with SELinux restrictions ast_bind to
+ a port reserved for another program by SELinux causes errno ==
+ EACCES. This caused random failures when binding rtp or udptl
+ sockets. Treat EACCES as a non-fatal error, try next port.
+ (closes issue ASTERISK-23134) Reported by: Corey Farrell
+
+2014-01-29 00:36 +0000 [r406860] Russell Bryant <russell at russellbryant.com>
+
+ * configs/queues.conf.sample: queues.conf.sample Fix documented
+ default for persistentmembers Closes issue ASTERISK-22662
+
+2014-01-28 23:02 +0000 [r406801] Kevin Harwell <kharwell at digium.com>
+
+ * cel/cel_radius.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, cdr/cdr_radius.c: cdr_radius, cel_radius: build
+ agains libfreeradius-client Asterisk's RADIUS module currently
+ build against libradiusclient-ng, but this project has been
+ superseeded by libfreeradius-client. The API is 99% compatible
+ except that the header name has changed, the library name has
+ changed, and the configuration file location has changed. (closes
+ issue ASTERISK-22980) Reported by: Jeremy Lainé Patches:
+ freeradius-client.patch uploaded by sharky (license 6561)
+
+2014-01-28 16:36 +0000 [r406721] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/rtp_engine.c: rtp_engine: improved handling of get_rtp_info
+ failure In ast_rtp_instance_make_compatible(), after a failure of
+ channel tech call get_rtp_info() to return peer_instance, the
+ null pointer would be passed to ao2_ref, producing an error that
+ looked like a refernce counting problem but is not. This patch
+ corrects that and adds helpful LOG_ERROR messages to indicate
+ which failure path occurred. (issue AST-1276) Review:
+ https://reviewboard.asterisk.org/r/3156/
+
+2014-01-27 20:34 +0000 [r406566-406643] Russell Bryant <russell at russellbryant.com>
+
+ * main/config.c: Allow nested #includes in extconfig.conf
+ extconfig.conf was hard-coded to not allow nested includes for
+ some reason. The code has been this way since a patch was merged
+ for ASTERISK-3333 (revision 4889), which was a significant update
+ to this code ("Merge config updates"). I can't figure out any
+ good reason why this should be limited. This patch just removes
+ the limit and uses the default nesting depth limit. Closes issue
+ ASTERISK-17837 Review: https://reviewboard.asterisk.org/r/3159/
+
+ * main/file.c, include/asterisk/channel.h, main/channel.c: Protect
+ ast_filestream object when on a channel The ast_filestream object
+ gets tacked on to a channel via chan->timingdata. It's a
+ reference counted object, but the reference count isn't used when
+ putting it on a channel. It's theoretically possible for another
+ thread to interfere with the channel while it's unlocked and
+ cause the filestream to get destroyed. Use the astobj2 reference
+ count to make sure that as long as this code path is holding on
+ the ast_filestream and passing it into the file.c playback code,
+ that it knows it's valid. Bug reported by Leif Madsen. Review:
+ https://reviewboard.asterisk.org/r/3135/
+
+2014-01-26 22:59 +0000 [r406514] Richard Mudgett <rmudgett at digium.com>
+
+ * main/tcptls.c: tcptls.c: Add missing cleanup on off nominal path.
+
+2014-01-24 22:56 +0000 [r406417] Richard Mudgett <rmudgett at digium.com>
+
+ * main/cel.c: CEL: Protect data structures during reload and
+ shutdown. The CEL data structures need to be protected during a
+ configuration reload and shutdown. Asterisk crashed during a
+ shutdown because CEL events were still in flight and the CEL data
+ structures were already destroyed. * Protected the appset and
+ linkedids ao2 containers using the reload_lock. * Added NULL
+ checks before use of the appset and linkedids ao2 containers in
+ case the CEL module is already shutdown. * Fixed overloading of
+ the linkedids held objects reference count. During shutdown any
+ held objects would be leaked. * Fixed memory leak of linkedids
+ held objects if the LINKEDID_END is not being tracked. The
+ objects in the linkedids container were not removed if the
+ LINKEDID_END event is not used. * Added access protection to the
+ appset container during the CLI "cel show status" command. * Made
+ CEL config reload not set defaults if the cel.conf file is
+ invalid. (closes issue AST-1253) Reported by: Guenther Kelleter
+ Review: https://reviewboard.asterisk.org/r/3127/
+
+2014-01-24 20:57 +0000 [r406360] Jonathan Rose <jrose at digium.com>
+
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