[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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+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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