[asterisk-commits] bebuild: tag 1.8.29.0-rc1 r418175 - /tags/1.8.29.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jul 8 09:40:45 CDT 2014


Author: bebuild
Date: Tue Jul  8 09:40:42 2014
New Revision: 418175

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=418175
Log:
Importing files for 1.8.29.0-rc1 release.

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+2014-07-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.28.0-rc1 Released.
+
+2014-07-03 21:38 +0000 [r417956]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.c,
+	  configs/chan_dahdi.conf.sample, UPGRADE.txt: chan_dahdi: Add
+	  inband_on_setup_ack compatibility option. The new
+	  inband_on_setup_ack option causes Asterisk to assume inband audio
+	  may be present when a SETUP_ACKNOWLEDGE message is received.
+	  Q.931 Section 5.1.3 says that in scenarios with overlap dialing,
+	  when a dialtone is sent from the network side, progress indicator
+	  8 "Inband info now available" MAY be sent to the CPE if no digits
+	  were received with the SETUP. It is thus implied that the ie is
+	  mandatory if digits came with the SETUP and dialtone is needed.
+	  This option should be enabled, when the network sends dialtone
+	  and you want to hear it, but the network doesn't send the
+	  progress indicator when needed. NOTE: For Q.SIG setups this
+	  option should be enabled when outgoing overlap dialing is also
+	  enabled because Q.SIG does not send the progress indicator with
+	  the SETUP ACK. The commit -r413714 (AST-1338) which causes this
+	  issue was dealing with a SIP-to-ISDN interoperability issue. This
+	  commit is a merge of the two patches indicated below.
+	  ASTERISK-23897 #close Reported by: Pavel Troller Patches:
+	  pri-4.diff (license #6302) patch uploaded by Pavel Troller
+	  jira_asterisk_23897_v11.patch (license #5621) patch uploaded by
+	  rmudgett Review: https://reviewboard.asterisk.org/r/3633/
+
+2014-07-03 11:19 +0000 [r417797]  Matthew Jordan <mjordan at digium.com>
+
+	* main/utils.c: main/untils: Prevent potential infinite loop in
+	  ast_careful_fwrite A loop in ast_careful_fwrite exists that will
+	  continually attempt to write to a file stream, even in the
+	  presence of EAGAIN/EINTR errors. However, if a connection that
+	  uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
+	  call to fflush may return EAGAIN/EINTER along with EOF. A
+	  subsequent call to fflush will return EOF but not clear errno,
+	  resulting in an infinite loop. This patch clears errno after it
+	  is detected and handled the loop, such that any subsequent call
+	  to fflush will not get erroneously stuck. Review:
+	  https://reviewboard.asterisk.org/r/3704 ASTERISK-23984 #close
+	  Reported by: Steve Davies patches: fflush_loop_fix uploaded by
+	  one47 (License 5012)
+
+2014-06-30 03:20 +0000 [r417587]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_sip.c: chan_sip: be more tolerant of whitespace
+	  between attributes in SDP fmtp line This patch is essentially a
+	  backport of a small portion of r397526 from ASTERISK-21981. In
+	  that patch, pass through support and format attribute negotiation
+	  was added for Opus. Part of that included being more tolerant to
+	  whitespace in the fmtp line of an SDP; that part of the patch is
+	  being applied here. As the author of the backport pointed out, in
+	  SDP, the fmtp line is allowed to include whitespace between
+	  attributes. RFC 3267 chapter 8.3 (from 2001) includes an example
+	  for this. This was not removed in the updated RFC 4867 in 2007.
+	  Note that this patch only applies to audio in Asterisk 1.8, which
+	  is a bit more limited in its support for format attributes. It
+	  does have limited support for some codecs, so this patch is still
+	  useful in this version. Review:
+	  https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 Reported
+	  by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch
+	  uploaded by Alexander Traud (License 6520)
+
+2014-06-27 19:24 +0000 [r417480-417500]  Corey Farrell <git at cfware.com>
+
+	* main/astobj2.c: Ensure REF_DEBUG records entrys for attempts to
+	  ao2_ref an invalid object This change ensures that
+	  __ao2_ref_debug writes to ref_log when given a non-NULL pointer
+	  to an invalid ao2 object. This is to ensure that we record any
+	  attempt manipulate references of already freed objects.
+	  ASTERISK-23948 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3677/
+
+	* contrib/scripts/refcounter.py: refcounter.py: prevent use of
+	  excessive RAM with large refs logs When processing a 212MB refs
+	  file, refcounter.py used over 3GB of RAM. This change greatly
+	  reduces memory usage in two ways: * Saving object history in
+	  whole lines instead of separated values. * Not saving
+	  normal/skewed/leaked object lists unless they are requested.
+	  ASTERISK-23921 #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3668/
+
+2014-06-26 12:21 +0000 [r417318]  Matthew Jordan <mjordan at digium.com>
+
+	* main/udptl.c: udptl: Correct FEC to not consider negative
+	  sequence numbers as missing When using FEC, with span=3 and
+	  entries=4 Asterisk will attempt to repair the packet with
+	  sequence number 5, as it will see that packet -4 is missing. The
+	  result is Asterisk sending garbage packets that can kill a fax.
+	  This patch adds a check to see if the sequence number is valid
+	  before checking if the packet is missing. Review:
+	  https://reviewboard.asterisk.org/r/3657/ ASTERISK-23908 #close
+	  Reported by: Torrey Searle patches: udptl_fec.patch uploaded by
+	  Torrey Searle (License 5334)
+
+2014-06-26 10:02 +0000 [r417248]  Corey Farrell <git at cfware.com>
+
+	* channels/chan_sip.c: chan_sip: Fix handling of "From" headers
+	  longer than 256 characters From headers were processed using a
+	  256 character buffer on the stack. This change replaces that with
+	  a heap allocation by ast_strdup. ASTERISK-23790 #close Reported
+	  by: uniken1 Tested by: uniken1 Review:
+	  https://reviewboard.asterisk.org/r/3669/ Patches:
+	  chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes
+	  (license 5674)
+
+2014-06-23 14:34 +0000 [r417076]  Rusty Newton <rnewton at digium.com>
+
+	* configs/features.conf.sample: main/features - documentation -
+	  reformat examples and options in features.conf.sample to show
+	  clearly which options apply in which section The features.conf
+	  sample can be a bit confusing about what parking options can be
+	  set only in the general context, or both in the general context
+	  (for the default parking lot) and in other parking lot contexts.
+	  A bug was filed due to confusion and a little googling will show
+	  lots of other confused users. Despite some comments on the
+	  individual options, it still reads in a confusing way. In this
+	  patch I separate out those options with some headings in to
+	  attempt a better layout. I went ahead and modified other headings
+	  in the file, or added them to facilitate better visual scanning.
+	  ASTERISK-23667 #close Review:
+	  https://reviewboard.asterisk.org/r/3621/
+
+2014-06-22 20:46 +0000 [r417016]  George Joseph <george.joseph at fairview5.com>
+
+	* Makefile.rules, Makefile: build: Turn FORTIFY_SOURCE off if
+	  DONT_OPTIMIZE is set. AST_FORTIFY_SOURCE is automatically set in
+	  ./Makefile even if DONT_OPTIMIZE is set in menuselect. This
+	  causes gcc to complain that _FORTIFY_SOURCE requires optimization
+	  and the build will fail. You can specify "make
+	  AST_FORTIFY_SOURCE=''" but I always forget. This patch moves the
+	  set of AST_FORTIFY_SOURCE to Makefile.rules and only sets it if
+	  DONT_OPTIMIZE is "no". The move is necessary because the
+	  top-level Makefile doesn't include menuselect.makeopts. This
+	  doesn't solve the entire problem however because res_config_mysql
+	  seems to force _FORTIFY_SOURCE so res_config_mysql has to be
+	  disabled for now if DONT_OPTIMIZE is set. Tested by: George
+	  Joseph Review: https://reviewboard.asterisk.org/r/3664/
+
+2014-06-20 23:12 +0000 [r416869-416929]  George Joseph <george.joseph at fairview5.com>
+
+	* configure, include/asterisk/autoconfig.h.in: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. A regenerated ./configure and
+	  include/asterisk/autoconfig.h.in are included but can be
+	  regenerated by running ./bootstrap.sh at any time. Tested by:
+	  George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/
+
+	* autoconf/ast_ext_tool_check.m4: build: Allow
+	  autoconf/ast_ext_tool_check to handle cross-compiling better.
+	  ast_ext_tool_check.m4 isn't handling cases where a path to a
+	  package is provided (E.G. --with-mysqlclient=/some/sysroot) and
+	  the package has a config tool (E.G. mysql_config) and the package
+	  has its own subdirectories in include or lib. For example,
+	  mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql but
+	  ast_ext_tool_check sets MYSQLCLIENT_LIB to
+	  ${MYSQLCLIENT_DIR}/usr/lib. libxml2 has the same problem with its
+	  includes. They're in ${LIBXML2_DIR}/usr/include/libxml2 not
+	  directly in ${LIBXML2_DIR}/usr/include. Both cause configure to
+	  fail and there are others in the same boat. The problem is caused
+	  by logic in ast_ext_tool_check that overrides the result of the
+	  config tool's --cflags and --libs options if package_DIR is set.
+	  This patch prepends package_DIR (if specified) to the -L and -I
+	  results from the package's config tool instead of overriding
+	  them. Tested by: George Joseph Tested by: Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3550/
+
+2014-06-19 19:33 +0000 [r416732]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/sip/reqresp_parser.c, main/test.c: Fix build warnings
+	  with TEST_FRAMEWORK enabled
+
+2014-06-19 15:59 +0000 [r416578-416667]  George Joseph <george.joseph at fairview5.com>
+
+	* pbx/pbx_lua.c: Remove the problematic and unneeded
+	  AST_MODFLAG_GLOBAL_SYMBOLS from pbx_lua.c
+	  AST_MODFLAG_GLOBAL_SYMBOLS was causing the module to be
+	  incorrectly loaded before pbx_config. pbx_config was therefore
+	  blowing away contexts that were created by pbx_lua. With
+	  AST_MODFLAG_DEFAULT the load order is now correct and contexs are
+	  being properly merged. AST_MODFLAG_GLOBAL_SYMBOLS was not needed
+	  anyway since no other modules needed its global symbols that
+	  early. ASTERISK-23818 #close Reported by: Dennis Guse Tested by:
+	  Dennis Guse Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3629/
+
+	* configs/extensions.lua.sample: Update extensions.lua.sample with
+	  naming conflict guidance. The sample extensions.lua was causing
+	  pbx_lua to fail to load when parsing 'app.goto("default", "s",
+	  1)' because in Lua 5.2, 'goto' is now a reserved word. This patch
+	  adds guidance to extensions.lua.sample and changed
+	  'app.goto("default", "s", 1)' to 'app.['goto']("default", "s",
+	  1)'. https://reviewboard.asterisk.org/r/3627/ ASTERISK-23844
+	  #comment This commit fixes 1.8, patch for 11->trunk coming.
+
+2014-06-17 18:22 +0000 [r416500]  Mark Michelson <mmichelson at digium.com>
+
+	* funcs/func_strings.c: Allow the PUSH and UNSHIFT functions to set
+	  inheritable channel variables.
+
+2014-06-17 16:20 +0000 [r416439]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_musiconhold.c: MoH: Don't restart stream on repeated
+	  start calls Currently, music on hold will stop and then start
+	  again from the beginning if ast_moh_start() is called multiple
+	  times. This can happen if a call is put on hold repeatedly (the
+	  channel receives multiple HOLD control frames) and can be
+	  triggered from ARI by starting MoH on a channel multiple times.
+	  This is fairly jarring/annoying to users. This change prevents
+	  MoH from being restarted if the requested music class is the same
+	  as the one currently playing. This includes an extra check to
+	  prevent the errors previously experienced in the testsuite and
+	  has 100+ test runs behind it. Review:
+	  https://reviewboard.asterisk.org/r/3615/
+
+2014-06-16 08:52 +0000 [r416336]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* cdr/cdr_sqlite3_custom.c, cel/cel_sqlite3_custom.c: We have faced
+	  situation when using CDR and CEL by sqlite3 modules. With system
+	  having high load (~100 concurrent calls created by sipp) we found
+	  many cdr and cel records missed. There is special finction in
+	  sqlite3, that make able to fix this situation -
+	  sqlite3_wait_timeout, that also can replace awful code
+	  cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be
+	  used for aastdb and res_config_sqlite3 to avoid missed writes to
+	  sqlite db. #ASTERISK-23766 #close Reported by: Igor Goncharovsky
+	  Review: https://reviewboard.asterisk.org/r/3559/
+
+2014-06-15 21:16 +0000 [r416251]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_musiconhold.c: MoH: Undo commit r416150 (1.8) This patch
+	  reverts r416150. When the comparison between mohclass->name and
+	  state->class->name is made, you are not guaranteed that (a)
+	  state->class is non-NULL or that state or state->class are in a
+	  safe state. Crashes caught by the bridges/transfer_capabilities
+	  test.
+
+2014-06-13 13:03 +0000 [r416150]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_musiconhold.c: MoH: Don't restart stream on repeated
+	  start calls Currently, music on hold will stop and then start
+	  again from the beginning if ast_moh_start() is called multiple
+	  times. This can happen if a call is put on hold repeatedly (the
+	  channel receives multiple HOLD control frames) and can be
+	  triggered from ARI by starting MoH on a channel multiple times.
+	  This is fairly jarring/annoying to users. This change prevents
+	  MoH from being restarted if the requested music class is the same
+	  as the one currently playing. Review:
+	  https://reviewboard.asterisk.org/r/3615/
+
+2014-06-13 04:58 +0000 [r416066]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/tcptls.c, main/manager.c, channels/chan_sip.c, main/http.c,
+	  include/asterisk/tcptls.h: AST-2014-007: Fix of fix to allow AMI
+	  and SIP TCP to send messages. ASTERISK-23673 #close Reported by:
+	  Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/
+
+2014-06-12 21:15 +0000 [r415998]  Rusty Newton <rnewton at digium.com>
+
+	* main/pbx.c: main/pbx - documentation - enhance 'core show hints'
+	  and 'core show hint' help text Adds descriptive help text to
+	  'core show hints' and 'core show hint'. The text describes the
+	  various columns for the sake of clarity. ASTERISK-23764 Review:
+	  https://reviewboard.asterisk.org/r/3610/
+
+2014-06-12 17:16 +0000 [r415908]  Corey Farrell <git at cfware.com>
+
+	* channels/sip/sdp_crypto.c: chan_sip: DEBUG messages in
+	  sdp_crypto.c display despite a DEBUG level of zero Change debug
+	  level for messages in sdp_crypto.c from zero to one. This ensures
+	  the messages are not displayed when debugging is disabled. Change
+	  does not apply to 12+ as it was already fixed in those versions.
+	  ASTERISK-23246 #close Reported by: Rusty Newton Review:
+	  https://reviewboard.asterisk.org/r/3605/
+
+2014-06-12 16:05 +0000 [r415841]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/tcptls.h, configs/http.conf.sample,
+	  include/asterisk/utils.h, main/tcptls.c, main/manager.c,
+	  channels/chan_sip.c, main/http.c, UPGRADE.txt, main/utils.c:
+	  AST-2014-007: Fix DOS by consuming the number of allowed HTTP
+	  connections. Simply establishing a TCP connection and never
+	  sending anything to the configured HTTP port in http.conf will
+	  tie up a HTTP connection. Since there is a maximum number of open
+	  HTTP sessions allowed at a time you can block legitimate
+	  connections. A similar problem exists if a HTTP request is
+	  started but never finished. * Added http.conf session_inactivity
+	  timer option to close HTTP connections that aren't doing
+	  anything. Defaults to 30000 ms. * Removed the undocumented
+	  manager.conf block-sockets option. It interferes with TCP/TLS
+	  inactivity timeouts. * AMI and SIP TLS connections now have
+	  better authentication timeout protection. Though I didn't remove
+	  the bizzare TLS timeout polling code from chan_sip. * chan_sip
+	  can now handle SSL certificate renegotiations in the middle of a
+	  session. It couldn't do that before because the socket was
+	  non-blocking and the SSL calls were not restarted as documented
+	  by the OpenSSL documentation. * Fixed an off nominal leak of the
+	  ssl struct in handle_tcptls_connection() if the FILE stream
+	  failed to open and the SSL certificate negotiations failed. The
+	  patch creates a custom FILE stream handler to give the created
+	  FILE streams inactivity timeout and timeout after a specific
+	  moment in time capability. This approach eliminates the need for
+	  code using the FILE stream to be redesigned to deal with the
+	  timeouts. This patch indirectly fixes most of ASTERISK-18345 by
+	  fixing the usage of the SSL_read/SSL_write operations.
+	  ASTERISK-23673 #close Reported by: Richard Mudgett
+
+2014-06-12 15:38 +0000 [r415833]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* apps/app_queue.c: app_queue: delayed state can cause early
+	  leavewhenempty ringing In app_queue, device state changes arrive
+	  in event messages and update the queue member status value. That
+	  value is checked in get_member_status() to decide that the caller
+	  should leave when there are no available members. Although event
+	  messages can be delayed by other activity, there is no adverse
+	  affect by lagged status except in one specific case: there is
+	  only one available member, it was just rung, and leavewhenempty
+	  is enabled set for ringing members. This change adds a direct
+	  check of the device state only under this condition where the
+	  caller may be dropped incorrectly, resolving this issue without
+	  affecting performance of app_queue normally. AST-1248 #close
+	  Review: https://reviewboard.asterisk.org/r/3595/ Reported by:
+	  Thomas Arimont
+
+2014-06-10 09:11 +0000 [r415598]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: chan_ooh323: fix loading module failure if
+	  there no accessible h323_log or ooh323 config file change return
+	  1 to return AST_MODULE_LOAD_FAILURE ASTERISK-23814 #close (closes
+	  issue ASTERISK-23814) Reported by: Igor Goncharovsky Patches:
+	  ASTERISK-23814-ast18.patch
+
+2014-06-09 11:55 +0000 [r415521]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* contrib/scripts/safe_asterisk: safe_asterisk: Cleanup additions
+	  to r415132. Replaced a stray echo that should've been a message
+	  call in safe_asterisk. I'm using the contents of the old message
+	  inside the if $NOTIFY so peoples log parsing scripts won't get
+	  confused by new messages. I'll clean that up in trunk. (Note that
+	  a 'make install' still won't overwrite your old safe_asterisk if
+	  it exists. See ASTERISK-21965.) ASTERISK-23492 #close
+
+2014-06-09 03:43 +0000 [r415463]  Corey Farrell <git at cfware.com>
+
+	* main/asterisk.c, include/asterisk.h, main/autoservice.c:
+	  autoservice: stop thread on graceful shutdown This change adds
+	  thread shutdown to autoservice for graceful shutdowns only.
+	  ast_register_cleanup is backported to 1.8 to allow this. The
+	  logger callid is also released on shutdown in 11+. ASTERISK-23827
+	  #close Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3594/
+
+2014-06-06 21:13 +0000 [r415359]  Jonathan Rose <jrose at digium.com>
+
+	* main/config.c, main/manager.c, channels/chan_sip.c,
+	  include/asterisk/config.h, include/asterisk/manager.h: chan_sip:
+	  Fix order of variables specified in SIPNotify action Prior to
+	  this patch, sequential variables would be ordered in reverse from
+	  the order specified in the manager action. Review:
+	  https://reviewboard.asterisk.org/r/3588/
+
+2014-06-05 17:36 +0000 [r415225]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/config.c: config: Fix config files not reloading when only
+	  an included file changes. The twisted logic determining if a
+	  config file should be reloaded was mostly broken and disabled.
+	  The incorrect test that ASTERISK-23383 fixed actually reenabled
+	  the broken logic. The incorrect test was causing the timestamp to
+	  always be cleared which caused config files with includes to
+	  always be reloaded. * Made wildcard includes always cause a
+	  reload. Determining if a file was deleted cannot be determined
+	  without restructuring the cache to determine if any files are
+	  missing from the last files actually loaded. Also without
+	  refactoring config_text_file_load(), the glob loop couldn't check
+	  more than one file for changes anyway. * Made remove the cache
+	  entry if the file no longer exists when trying to get its
+	  timestamp or it is no longer a regular file. This fixes the
+	  corner case where the file was loaded, then deleted, then the
+	  config reloaded, then the file restored with the same timestamp,
+	  and then the config reloaded again. * Made remove the cache entry
+	  include list when actually loading the file. This gets rid of any
+	  stale includes the file had from the last time the file was
+	  loaded. ASTERISK-23683 #close Reported by: tootai Review:
+	  https://reviewboard.asterisk.org/r/3575/
+
+2014-06-04 15:16 +0000 [r415132]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* contrib/scripts/safe_asterisk: safe_asterisk: Cleanup and debian
+	  compatibility. Cleans up the safe_asterisk script and adds the
+	  ASTSAFE_FOREGROUND option that allows the debian asterisk init
+	  script to capture the right pid. * Drop the vim #modeline which
+	  wasn't used. Use test consistently without the odd configure xno
+	  syntax. Double quote all paths. General cleanup. * Don't output
+	  message()s to the console but only to TTY if set. * Allow TTY to
+	  be "no" as well as empty (debian compatibility with
+	  debian/patches/safe_asterisk-config). * Add option to export
+	  ASTSAFE_FOREGROUND=1 from the init script that calls this to
+	  disable backgrounding. Debian uses a similar method in
+	  debian/patches/safe_asterisk-nobg). ASTERISK-23492 #close Review:
+	  https://reviewboard.asterisk.org/r/3574/
+
+2014-06-04 07:18 +0000 [r415060]  Corey Farrell <git at cfware.com>
+
+	* apps/app_confbridge.c: app_confbridge: Correct verification of
+	  conference name length Conference names were not checked for
+	  maximum length, allowing unexpected behaviour. This change adds
+	  checking to ensure the maximum length is not exceeded. The
+	  maximum length is also changed from 32 to AST_MAX_EXTENSION.
+	  ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki
+	  Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by
+	  coreyfarrell (license 5909) confbridge-enforce_max-11up.patch
+	  uploaded by coreyfarrell (license 5909)
+
+2014-06-03 07:31 +0000 [r414997]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* funcs/func_odbc.c: func_odbc: Fix fixed size buffers fix
+	  (r414968). The change that removed the fixed size buffers in
+	  odbc-related code -- removing arbitrary column width limits --
+	  was incomplete. This change adds: no segfault on writesql without
+	  insertsql and return value checks after strdup. While I was in
+	  the vicinity I cleaned up the linefeeds in the odbc function
+	  descriptions, moved some code for clarity, removed some blobs and
+	  noted (but didn't fix) that the 'odbc write ... exec' CLI command
+	  doesn't behave as the dialplan equivalent when insertsql= is
+	  used. ASTERISK-23582 #close Review:
+	  https://reviewboard.asterisk.org/r/3579/
+
+2014-05-30 11:50 +0000 [r414880]  Matthew Jordan <mjordan at digium.com>
+
+	* main/config.c: main/config.c: AMI action UpdateConfig EmptyCat
+	  clears all categories When invoking UpdateConfig AMI action with
+	  Action set to EmptyCat, Asterisk will make all categories empty
+	  in the config but the one requested with a Cat variable. This is
+	  due to a bug in ast_category_empty (main/config.c) that makes an
+	  incorrect comparison for a category name. This patch corrects the
+	  comparison such that only the requested category is cleared.
+	  Review: https://reviewboard.asterisk.org/r/3573/ ASTERISK-23803
+	  #close Reported by: zvision patches: manager.c.diff uploaded by
+	  zvision (License 5755)
+
+2014-05-29 15:55 +0000 [r414813]  Kinsey Moore <kmoore at digium.com>
+
+	* main/pbx.c: PBX: Prevent incorrect hint parsing Dynamic and
+	  pattern matching hints should not be checked for their last known
+	  state until they are instantiated by subscribers. (closes issue
+	  AFS-56) Reported by: John Hardin Patch AFS-56-pbx.diff submitted
+	  by Matt Jordan (license 6283)
+
+2014-05-28 11:34 +0000 [r414693]  Joshua Colp <jcolp at digium.com>
+
+	* funcs/func_odbc.c, res/res_config_odbc.c: res_config_odbc: Use
+	  dynamically sized buffers to store row data so values do not get
+	  truncated. ASTERISK-23582 #close ASTERISk-23582 #comment Reported
+	  by: Walter Doekes Review:
+	  https://reviewboard.asterisk.org/r/3557/
+
+2014-05-27 21:16 +0000 [r414564-414620]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: chan_sip: Start session timer at 200, not at
+	  INVITE. Asterisk started counting the session timer at INVITE
+	  while the other end correctly started at 200. This meant that for
+	  short session-expiries (90 seconds) combined with long ringing
+	  times (e.g. 30 seconds), asterisk would wrongly assume that the
+	  timer was hit before the other end thought it was time to send a
+	  session refresh. This resulted in prematurely ended calls. This
+	  changes the session timer to start counting first at 200 like RFC
+	  says it should. (Also removed a few excess NULL checks that would
+	  never hit, because if they did, asterisk would have crashed
+	  already.) ASTERISK-22551 #close Reported by: i2045 Review:
+	  https://reviewboard.asterisk.org/r/3562/
+
+	* res/res_config_odbc.c: res_config_odbc: Fix old and new
+	  ast_string_field memory leaks. The ODBC realtime driver uses ^NN
+	  parameter encoding to cope with the special meaning of the
+	  semi-colon. A semi-colon in a field is interpreted as if the key
+	  was supplied twice, something which isn't otherwise possible with
+	  fixed database columns. E.g. allow=alaw;ulaw is parsed as
+	  allow=alaw and allow=ulaw. A literal semi-colon is rewritten to
+	  ^3B when stored in the database. The module uses a stringfield to
+	  efficiently store the encoded parameters. However, this
+	  stringfield wasn't always freed in some off-nominal cases. Commit
+	  r413241 fixed initialization so the encoding for INSERT and
+	  DELETE queries wouldn't crash. (Only SELECTs and UPDATEs worked
+	  apparently.) But that commit forgot the frees. This change cleans
+	  that up. Review: https://reviewboard.asterisk.org/r/3555/
+
+2014-05-23 16:06 +0000 [r414488]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_sip.c: Backport Asterisk 11 r413876 to 1.8 ........
+	  r413876 | jrose | 2014-05-13 12:40:00 -0500 (Tue, 13 May 2014) |
+	  6 lines chan_sip: Add TLS and SRTP status to CLI command 'sip
+	  show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock
+	  Review: https://reviewboard.asterisk.org/r/3474/
+
+2014-05-29  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.28.0 Released.
+
+2014-05-22  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.28.0-rc1 Released.
+
+2014-05-22 15:47 +0000 [r414401]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_meetme.c: app_meetme: Don't interrupt MOH for waitmarked
+	  users. Occasionally, when the last marked user leaves the
+	  conference, waitmarked users don't get MOH if MOH is supposed to
+	  be played while a waitmarked user is waiting for another marked
+	  user. * Made not interrupt MOH when the user is a waitmarked
+	  user. The waitmarked user doesn't need to hear any leave
+	  announcements from the conference as the user would have already
+	  heard different leave announcements if they were enabled.
+	  Apparently DAHDI occasionally sends unending non-silent streams
+	  to these users or a normal user still in the conference has
+	  continuous high background noise. These non-silent streams cause
+	  MOH to be suspended while the never ending "announcement" is
+	  played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
+	  by: Tyler Stewart Review:
+	  https://reviewboard.asterisk.org/r/3543/
+
+2014-05-22 13:58 +0000 [r414345]  Matthew Jordan <mjordan at digium.com>
+
+	* UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag
+
+2014-05-21 22:01 +0000 [r414269]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_local.c: chan_local: Only block media frames when a
+	  generator is on both ends of a local channel. The fix for
+	  ASTERISK-12292 was a bit too aggressive. You could have
+	  generators pointed at each other on local channels but need to
+	  get other kinds of frames such as DTMF or CONNECTED_LINE frames
+	  accross.
+
+2014-05-21 18:58 +0000 [r414214]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* funcs/func_strings.c: pbx.c: prevent potential crash from
+	  recursive replace() Recurisve usage of replace() resulted in
+	  corruption of the temporary string storage and potential crash.
+	  By changing the string to be allocated separtely per instance,
+	  this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
+	  Meer ASTERISK-23650 #close Review:
+	  https://reviewboard.asterisk.org/r/3539/
+
+2014-05-19 13:31 +0000 [r414152]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: chan_ooh323: fix h323_log full path name *
+	  fix to use astlogdir option for h323_log file instead of
+	  hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky
+	  Patches: ooh323_logger_patch.diff
+
+2014-05-16 20:00 +0000 [r413991-414067]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone detection.
+	  * Check if waitingfordt (waitfordialtone) is enabled in
+	  dahdi_read() to allow the DSP to operate early enough to detect
+	  dialtone. * Made use the correct variable in
+	  my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
+	  Davies Patches: dialtone_detect_fix (license #5012) patch
+	  uploaded by Steve Davies Review:
+	  https://reviewboard.asterisk.org/r/3534/
+
+	* apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI conference
+	  data structure. Starting a conference recording using the admin
+	  menu overwrites the DAHDI conference data structure used to
+	  modify the admin user's conference mute mode. * Made no longer
+	  pass the user's DAHDI conference data structure into the menu
+	  functions. The menu now uses its own DAHDI conference data
+	  structure to start the recording channel. * Moved the unlock
+	  conf->playlock to before playing the conf-full message. No sense
+	  keeping the lock while that prompt is playing. The user is never
+	  going to get into the conference at that point.
+
+2014-05-15 15:32 +0000 [r413949]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* apps/app_dial.c, channels/chan_local.c, UPGRADE.txt:
+	  chan_local+app_dial: Propagagate call answered elsewhere over
+	  local channels. AST_FLAG_ANSWERED_ELSEWHERE was not propagated
+	  back from local channels. It is now. That means that when a call
+	  is picked up from a callgroup of local channels, the other
+	  channels will now properly see it as "picked up". This occurs
+	  when you use a construct like
+	  Dial(Local/a at context&Local/b at context) where a at context and
+	  b at context dial two chan_sip devices respectively. If one device
+	  picks up, the other will not see "1 missed call" anymore. In this
+	  respect, it now behaves the same as when doing Dial(SIP/a&SIP/b).
+	  Review: https://reviewboard.asterisk.org/r/3540/
+
+2014-05-14 15:27 +0000 [r413894]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a few
+	  free()'s that should be ast_free()'s. Reverted an old workaround
+	  that isn't necessary. Reorder a tiny bit of code. Remove a bit of
+	  commented-out code. Review:
+	  https://reviewboard.asterisk.org/r/3536/
+
+2014-05-13 14:32 +0000 [r413787-413832]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* channels/chan_sip.c: chan_sip+CEL: Add missing ANSWER and PICKUP
+	  events to INVITE/w/replaces pickup. When doing a "BLF-style call
+	  pickup" -- an INVITE with Replaces: header -- the CEL log would
+	  lack the ANSWER and PICKUP events. This patch adds the two
+	  missing events to the handle_invite_replaces() function.
+	  ASTERISK-22977 #close Review:
+	  https://reviewboard.asterisk.org/r/3073/
+
+	* main/rtp_engine.c: rtp: Fix case typo in H263+ mime.
+	  http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
+	  canonical mime subtype is "H263-1998", not "h263-1998". Original
+	  code was added in r183101 on 2009-03-19 02:26:50 +0100. This
+	  fixes issues with Polycom phones. ASTERISK-23665 #close
+	  ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
+	  Maudoux, backported by me. Review:
+	  https://reviewboard.asterisk.org/r/3529/
+
+2014-05-12 23:08 +0000 [r413714]  Richard Mudgett <rmudgett at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
+	  PROGRESS events when overlap dialing is enabled. When overlap
+	  dialing is enabled, the lack of inband audio available
+	  information in the SETUP_ACKNOWLEDGE events causes an
+	  interoperability problem with SIP. sig_pri doesn't know if there
+	  is dialtone present when a SETUP_ACKNOWLEDGE is received so it
+	  assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
+	  SIP channel driver then sends out a 183 Session Progress and
+	  blocks the desired 180 Ringing message when the ALERTING message
+	  comes in. * Made the configure script detect if the installed
+	  version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
+	  Using the new API, made generate an AST_CONTROL_PROGRESS frame on
+	  an incoming SETUP_ACKNOWLEDGE message when the message indicates
+	  inband audio is present instead of assuming that dialtone is
+	  present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
+	  inband audio available indication only if dialtone is expected.
+	  The change also makes the fallback behaviour of sending the
+	  PROGRESS message better by sending it only if dialtone is
+	  expected. * Changed receiving a PROCEEDING message to not
+	  generate an AST_CONTROL_PROGRESS frame if the progress indication
+	  ie indicates non-end-to-end-ISDN. This helps interoperability
+	  with SIP. * Changed sending a PROCEEDING message in response to
+	  an AST_CONTROL_PROCEEDING frame to not indicate inband audio
+	  available. It was silly to do so anyway because the channel
+	  driver doesn't know if inband audio is even available. This helps
+	  interoperability with SIP. This patch and a corresponding change
+	  in libpri work together to allow Asterisk to control the inband
+	  audio available progress indication ie on the SETUP_ACKNOWLEDGE
+	  message when dialtone is present. AST-1338 #close Reported by:
+	  Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
+
+2014-05-09 23:02 +0000 [r413586-413592]  Kinsey Moore <kmoore at digium.com>
+
+	* funcs/func_env.c: Fix 32bit build for func_env
+
+	* channels/chan_sip.c: Fix 32bit build for chan_sip
+
+	* channels/chan_dahdi.c, channels/sig_analog.c,
+	  include/asterisk/astobj.h, main/event.c, funcs/func_iconv.c,
+	  channels/sip/config_parser.c, apps/app_stack.c, res/res_odbc.c,
+	  apps/app_adsiprog.c, res/res_calendar.c, main/udptl.c,
+	  main/stun.c, main/frame.c, channels/chan_sip.c,
+	  apps/app_festival.c, funcs/func_env.c, main/taskprocessor.c,
+	  channels/chan_iax2.c, apps/app_getcpeid.c, res/res_monitor.c,
+	  res/ael/pval.c, main/channel.c, main/manager.c,
+	  formats/format_pcm.c, funcs/func_srv.c, main/file.c,
+	  main/callerid.c, main/app.c, channels/chan_alsa.c, main/adsi.c,
+	  pbx/pbx_dundi.c, main/stdtime/localtime.c, res/res_fax_spandsp.c,
+	  main/sched.c, res/res_rtp_asterisk.c, cel/cel_pgsql.c,
+	  cdr/cdr_adaptive_odbc.c, res/res_musiconhold.c,
+	  channels/chan_gtalk.c, channels/sig_pri.c, res/res_srtp.c,
+	  main/io.c, channels/chan_jingle.c, channels/chan_phone.c,
+	  funcs/func_enum.c, res/res_config_odbc.c, apps/app_minivm.c,
+	  res/res_agi.c, main/features.c, apps/app_dumpchan.c,
+	  main/abstract_jb.c, main/logger.c, apps/app_sms.c,
+	  main/audiohook.c, pbx/pbx_config.c, main/bridging.c, main/dsp.c,
+	  apps/app_voicemail.c, apps/app_dial.c,
+	  res/res_calendar_exchange.c, main/security_events.c,
+	  res/res_fax.c, res/res_timing_dahdi.c, funcs/func_sysinfo.c,
+	  main/utils.c, main/devicestate.c, res/res_jabber.c,
+	  res/res_pktccops.c, main/cli.c, main/data.c, cel/cel_odbc.c,

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