[svn-commits] bebuild: tag 1.8.29.0-rc1 r418175 - /tags/1.8.29.0-rc1/
SVN commits to the Digium repositories
svn-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.
Added:
tags/1.8.29.0-rc1/.lastclean (with props)
tags/1.8.29.0-rc1/.version (with props)
tags/1.8.29.0-rc1/ChangeLog (with props)
Added: tags/1.8.29.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.29.0-rc1/.lastclean?view=auto&rev=418175
==============================================================================
--- tags/1.8.29.0-rc1/.lastclean (added)
+++ tags/1.8.29.0-rc1/.lastclean Tue Jul 8 09:40:42 2014
@@ -1,0 +1,1 @@
+40
Propchange: tags/1.8.29.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.29.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.29.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.29.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.29.0-rc1/.version?view=auto&rev=418175
==============================================================================
--- tags/1.8.29.0-rc1/.version (added)
+++ tags/1.8.29.0-rc1/.version Tue Jul 8 09:40:42 2014
@@ -1,0 +1,1 @@
+1.8.29.0-rc1
Propchange: tags/1.8.29.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.8.29.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.8.29.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.8.29.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.8.29.0-rc1/ChangeLog?view=auto&rev=418175
==============================================================================
--- tags/1.8.29.0-rc1/ChangeLog (added)
+++ tags/1.8.29.0-rc1/ChangeLog Tue Jul 8 09:40:42 2014
@@ -1,0 +1,48989 @@
+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 CÃvico Tested by: Iñaki
+ CÃvico 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,
[... 48307 lines stripped ...]
More information about the svn-commits
mailing list