[asterisk-commits] bebuild: tag 11.13.0-rc1 r423597 - /tags/11.13.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 19 15:20:25 CDT 2014
Author: bebuild
Date: Fri Sep 19 15:20:21 2014
New Revision: 423597
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423597
Log:
Importing files for 11.13.0-rc1 release.
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tags/11.13.0-rc1/.lastclean (with props)
tags/11.13.0-rc1/.version (with props)
tags/11.13.0-rc1/ChangeLog (with props)
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+2014-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.13.0-rc1 Released.
+
+2014-09-18 16:30 +0000 [r423400] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/astobj2.c, contrib/scripts/refcounter.py:
+ astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
+ Make astob2 REF_DEBUG output an invalid object line when an
+ invalid ao2 object ref/unref is attempted. This is similar to the
+ constructor/destructor lines. * Fixed refcounter.py to handle
+ skewed objects that have constructor/destructor states. * Made
+ refcounter.py highlight the invalid ao2 object refs by putting
+ them in their own section of the processed output file. * Made
+ refcounter.py highlight unreffing an object by more than one that
+ results in a negative ref count and the object being destroyed.
+ The abnormally destroyed object is reported in the invalid and
+ finalized object sections of the output. Review:
+ https://reviewboard.asterisk.org/r/3971/ ........ Merged
+ revisions 423349 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-18 16:19 +0000 [r423360] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: Properly handle cleanup
+ before starting FAXes. If faxing fails at a very early stage,
+ then it is possible for us to pass a NULL t30 state pointer to
+ spandsp, which spandsp is none too pleased with. This patch
+ ensures that we pass the correct pointer to spandsp in the
+ situation where we have not yet set our local t30 state pointer.
+ ASTERISK-24301 #close Reported by Matt Jordan Patches:
+ ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License
+ #5049)
+
+2014-09-18 14:42 +0000 [r423277] George Joseph <george.joseph at fairview5.com>
+
+ * main/config.c, main/manager.c, /, include/asterisk/config.h:
+ config: bug: Fix SEGV in ast_category_insert when matching
+ category isn't found If you call ast_category_insert with a match
+ category that doesn't exist, the list traverse runs out of 'next'
+ categories and you get a SEGV. This patch adds check for the
+ end-of-list condition and changes the signature to return an int
+ for success/failure indication instead of a void. The only
+ consumer of this function is manager and it was also changed to
+ use the return value. Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3993/ ........ Merged
+ revisions 423276 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-17 18:02 +0000 [r423150-423253] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Ensure that the thread
+ terminating pj stuff is registered.
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix 100% CPU usage due
+ to timer heap thread spinning. Side note: I need a vacation.
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix building when
+ pjproject is not used.
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix a myriad of TURN
+ client issues. 1. The number of file descriptors an ioqueue
+ instance can handle is fixed, so we now spawn the required number
+ to handle the load. 2. Our transport identifiers were exceeding
+ the range supported by pjnath. 3. The TURN client did not set up
+ client binding causing needless bandwidth usage. 4. The code no
+ longer updates address information on each packet. 5. STUN
+ traffic was getting looped back to Asterisk instead of going
+ through the TURN server. 6. Synchronization now ensures things
+ are completely setup or destroyed. 7. Logging now reflects the
+ target the TURN server is sending to/receiving from on our
+ behalf. ASTERISK-23577 #close Reported by: Jay Jideliov
+ ASTERISK-23634 #close Reported by: Roman Skvirsky Review:
+ https://reviewboard.asterisk.org/r/3982/
+
+2014-09-14 15:49 +0000 [r423067] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * configs/sip.conf.sample, /: chan_sip: Clarify that sipdebug=yes
+ cannot be undone by the CLI. Document it in sip.conf.
+ ASTERISK-24249 #close Reported by: Avinash Mohod Review:
+ https://reviewboard.asterisk.org/r/3926/ ........ Merged
+ revisions 423066 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-12 18:18 +0000 [r423010] Kinsey Moore <kmoore at digium.com>
+
+ * main/channel.c, /: Bridging: Fix bouncing native bridge This
+ fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge
+ could cause a bouncing native bridge. In the case of the
+ dial_LS_options test, this was a remote RTP bridge which caused
+ the audio path to continually cycle between Asterisk and the
+ remote endpoints generating a large number of SIP messages and
+ delaying the test long enough to cause it to fail (checking
+ timing was part of the test). The root cause was that the code to
+ decide whether to use native bridging was expecting a
+ time-remaining value of 0 to be the default instead of the actual
+ default value of -1. A value of 0 or negative numbers could also
+ be generated by preceding code in some circumstances. Both issues
+ are addressed in this patch. ASTERISK-24211 #close Reported by:
+ Matt Jordan Review: https://reviewboard.asterisk.org/r/3987/
+ ........ Merged revisions 423006 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-10 16:01 +0000 [r422903] George Joseph <george.joseph at fairview5.com>
+
+ * /, main/config.c: config: bug: fix truncation of included config
+ files on permissions error ast_config_text_file_save() currently
+ truncates include files as they are processed. If a subsequent
+ include file or the main config file has a permissions error that
+ prevents writing, earlier include files are left truncated
+ resulting in a frantic search for backups. This patch causes
+ ast_config_text_file_save to check for write access on all files
+ before it truncates any of them. Will be applied 1.8 > trunk.
+ Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3986/ ........ Merged
+ revisions 422900 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-07 00:08 +0000 [r422790] Rusty Newton <rnewton at digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile, /: Sounds/BuildSystem:
+ Modifications to include new releases and Japanese language.
+ Modifying Makefile and sounds.xml to include new core 1.4.26 and
+ extra 1.4.15 sound prompt releases, plus the new Japanese core
+ sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
+ Kevin McCoy Tested by: Rusty Newton ........ Merged revisions
+ 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-04 20:39 +0000 [r422625] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c, /: Manager: Require read permission for SYSTEM in
+ order to send FullyBooted Review:
+ https://reviewboard.asterisk.org/r/3969/ ........ Merged
+ revisions 422584 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-30 17:22 +0000 [r422440] George Joseph <george.joseph at fairview5.com>
+
+ * main/manager.c, /: manager: Make WaitEvent action respect
+ eventfilters A WaitEvent issued via an http session isn't
+ respecting eventfilters defined for the user. I just added a
+ match_filter to the predicate that controls astman_append. Tested
+ by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3958/ ........ Merged
+ revisions 422439 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-29 19:39 +0000 [r422294-422377] Matthew Jordan <mjordan at digium.com>
+
+ * doc/smsq.8 (added), /: doc: Add a manpage for the smsq utility
+ This patch adds a manpage for the smsq utility. Note that this is
+ one of the patches the Debian distro applies for the Asterisk
+ project, as per ASTERISK-24191. Review:
+ https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close
+ Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy
+ Laine (License 6561) ........ Merged revisions 422376 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * doc/aelparse.8 (added), /: doc: Add a manpage for the aelparse
+ utility This patch adds a manpage for the aelparse utility. Note
+ that this is one of the patches the Debian distro applies for the
+ Asterisk project, as per ASTERISK-24191. Review:
+ https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close
+ Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy
+ Laine (License 6561) ........ Merged revisions 422371 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, LICENSE: LICENSE: Clarify language in Asterisk's LICENSE to
+ allow for linking to UniMRCP The UniMRCP project distributes
+ Asterisk modules that integrate Asterisk with UniMRCP, and other
+ Asterisk users use the UniMRCP library as well. Unfortunately,
+ the UniMRCP license is Apache 2.0, which per the Free Software
+ Foundation, is not a compatible license with the GPLv2. "Please
+ note that this license is not compatible with GPL version 2,
+ because it has some requirements that are not in that GPL
+ version. These include certain patent termination and
+ indemnification provisions. The patent termination provision is a
+ good thing, which is why we recommend the Apache 2.0 license for
+ substantial programs over other lax permissive licenses." On the
+ other hand, UniMRCP is a great project and we'd like to let
+ people use it with Asterisk. This patch updates the LICENSE text
+ to allow users to link Asterisk with UniMRCP and distribute the
+ resulting binaries. ........ Merged revisions 422293 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-28 20:26 +0000 [r422274] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_iax2.c: chan_iax2: Fix Dynamic IAX2 Registrations
+ After Temporary DNS Failure The reporter on the issue found some
+ issues when upgrading from version 10 to 11 on 55 hosts. Two
+ situations that can occur with dynamic registrations. 1. With
+ dnsmgr disabled, if the host is not resolvable we are not trying
+ to resolve the host again when it is time to attempt to register
+ again. This results in never registering to the host. 2. With
+ dnsmgr enabled, when the host is temporarily not resolvable the
+ address is set to 0.0.0.0:0 and then when the host is resolvable
+ the port is not being restored and stays set to 0. This patch
+ resolves these two issues by: * Storing the hostname so that it
+ can be used for resolving with DNS. * Resolve the hostname on the
+ next scheduled attempt to register. * Storing the port used to
+ reach the host so that when the hostname is resolvable again, we
+ can set the port again if the port is still unset after looking
+ up the host. ASTERISK-23767 #close Reported by: David Herselman
+ Tested by: David Herselman, Michael L. Young Patches:
+ asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/3856/
+
+2014-08-27 15:01 +0000 [r422113] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_sip.c, tests/test_callerid.c (added),
+ tests/test_utils.c, main/callerid.c, main/utils.c,
+ include/asterisk/utils.h: CallerID: Fix parsing of malformed
+ callerid This allows the callerid parsing function to handle
+ malformed input strings and strings containing escaped and
+ unescaped double quotes. This also adds a unittest to cover many
+ of the cases where the parsing algorithm previously failed.
+ Review: https://reviewboard.asterisk.org/r/3923/ ........ Merged
+ revisions 422112 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-25 16:07 +0000 [r421977] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_musiconhold.c: res_musiconhold: Fix MOH restarting
+ where it left off from the last hold. Restore code removed by
+ https://reviewboard.asterisk.org/r/3536/ that introduced a
+ regression that prevents MOH from restarting were it left off the
+ last time. ASTERISK-24019 #close Reported by: Jason Richards
+ Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Review:
+ https://reviewboard.asterisk.org/r/3928/ ........ Merged
+ revisions 421976 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-24 17:19 +0000 [r421909] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Use the server reflexive ICE
+ candidate RTCP port as provided. This code originally worked
+ around an issue within res_rtp_asterisk itself. The wrong socket
+ was being used for the STUN check for RTCP, causing the port to
+ be the same as RTP. This was subsequently fixed and the RTCP port
+ provided for the ICE candidate is correct and does not need to be
+ incremented. ASTERISK-23997 #close Reported by: Badalian
+ Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav
+ (license 5249)
+
+2014-08-21 22:03 +0000 [r421800] Richard Mudgett <rmudgett at digium.com>
+
+ * /, res/res_musiconhold.c: res_musiconhold.c: Remove obsolete
+ REF_DEBUG code. Remove unneeded code that writes to the wrong
+ file location in an obsolete format. ........ Merged revisions
+ 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-21 21:00 +0000 [r421777] Jonathan Rose <jrose at digium.com>
+
+ * res/res_musiconhold.c, /: res_musiconhold: Fix reference leaks
+ caused when reloading with REF_DEBUG set Due to a faulty function
+ for debugging reference decrementing, it was possible to reduce
+ the refcount on the wrong object if two moh classes of the same
+ name were in the moh class container. (closes issue
+ ASTERISK-22252) Reported by: Walter Doekes Patches:
+ 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
+ 6182) ........ Merged revisions 398937 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-21 17:32 +0000 [r421718] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Don't use port derived from
+ fromdomain if it isn't set If a user does not provide a port in
+ the fromdomain setting, chan_sip will set the fromdomainport to
+ STANDARD_SIP_PORT (5060). The fromdomainport value will then get
+ used unilaterally in certain places. This causes issues with TLS,
+ where the default port is expected to be 5061. This patch
+ modifies chan_sip such that fromdomainport is only used if it is
+ not the standard SIP port; otherwise, the port from the SIP pvt's
+ recorded self IP address is used. Review:
+ https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close
+ Reported by: Elazar Broad patches: fromdomainport_fix.diff
+ uploaded by Elazar Broad (License 5835) ........ Merged revisions
+ 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-20 22:17 +0000 [r421602] Richard Mudgett <rmudgett at digium.com>
+
+ * /, main/cli.c: cli.c: Fix tab completion of "module load" when
+ MALLOC_DEBUG is enabled. filename_completion_function() returns
+ memory that was not allocated by the MALLOC_DEBUG allocation
+ tracker so the memory must be freed by ast_std_free(). ........
+ Merged revisions 421600 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-19 19:41 +0000 [r421443] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c, /: AMI Docs: Fix Status channel parameter
+ optionality ........ Merged revisions 421442 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-18 20:16 +0000 [r421328] George Joseph <george.joseph at fairview5.com>
+
+ * funcs/func_config.c, /: func_config: Change 'Not Found' message
+ from ERROR to DEBUG When you call the CONFIG dialplan function
+ with the name of a variable that doesn't exist in the target
+ context you get an ERROR. This does nothing but clutter up the
+ logs with messages that may be perfectly acceptable. Just because
+ a variable wasn't in the context doesn't mean it's an error.
+ Maybei t's optional or just needs to be defaulted or ignored.
+ This patch changes the log level from ERROR to DEBUG. If a
+ dialplan developer wants to debug their dialplan they still canby
+ setting the console debug level as needed. Tested by: George
+ Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........
+ Merged revisions 421327 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-17 23:07 +0000 [r421228-421233] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_dial.c, /: apps/app_dial: Fix Dial 'z' option The 'z'
+ option is supposed to disable the dial timeout in the case of a
+ call forward. Unfortunately, the wrong timeout timer was passed
+ to the do_forward function, resulting in the option not working.
+ ASTERISK-24225 #close Reported by: dimitripietro Tested by:
+ dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by
+ rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by
+ rmudgett (License 5621) ........ Merged revisions 421232 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, configure.ac: configure: Undefine FORTIFY_SOURCE
+ prior to defining it for patched gcc Some distributions of Linux
+ patch gcc to define FORTIFY_SOURCE when gcc is executed with
+ optimization. This "help" unfortunately results in re-definition
+ warnings when FORTIFY_SOURCE is later defined in Asterisk's build
+ system. This patch undefines FORTIFY_SOURCE prior to defining it
+ to prevent this warning. Review:
+ https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close
+ Reported by: Kilburn Tested by: Kilburn, wdoekes patches:
+ 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by
+ cloos (License 5956) 11.diff uploaded by cloos (License 5956)
+ 12.diff uploaded by cloos (License 5956) 13.diff uploaded by
+ cloos (License 5956) ........ Merged revisions 421227 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-15 15:36 +0000 [r421060-421164] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, main/app.c: app_voicemail/app: Remove test
+ events that were duplicated by r421059 Moving the test event
+ raised when a file is played back (which occurred in r421059)
+ broke the ever loving snot out of the voicemail tests. This
+ caused duplicate test events to get raised, as app_voicemail and
+ main/app were raising events prior to call ast_streamfile. The
+ voicemail tests did not enjoy getting multiple events. Since
+ raising the playback event in ast_streamfile is far more useful
+ to the vast majority of tests, this patch keeps the call there
+ and simply removes the extraneous calls that duplicated the
+ event. ........ Merged revisions 421125 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/file.c, main/app.c: main/file: Move test event to emit
+ PLAYBACK event more consistently This is being done in advance of
+ the test for ASTERISK-23953 ........ Merged revisions 421059 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-13 07:47 +0000 [r420897] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/logger.c: logger: Don't store verbose-magic in the log
+ files. In r399267, the verbose2magic stuff was edited. This time
+ it results in magic characters in the log files for multiline
+ messages. In trunk (and 13) this was fixed by the "stripping" of
+ those characters from multiline messages (in r414798). This is a
+ backport of that fix to 11. That fix is altered to actually strip
+ the characters and not replace them with blanks. Review:
+ https://reviewboard.asterisk.org/r/3901/ Review:
+ https://reviewboard.asterisk.org/r/3902/
+
+2014-08-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.12.0 Released.
+
+2014-08-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.12.0-rc1 Released.
+
+2014-08-11 10:36 +0000 [r420655-420715] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * /, main/utils.c: general: Fix memory Corruption in
+ __ast_string_field_ptr_build_va. If the space left in a
+ stringfield is between 0 and
+ (alignof(ast_string_field_allocation)-1) adding new data would
+ cause memory corruption, because we would assume enough space
+ (unsigned underrun). Thanks Arnd Schmitter for reporting and
+ finding out the cause! ASTERISK-23508 #close Reported by: Arnd
+ Schmitter Tested by: Arnd Schmitter, JoshE Review:
+ https://reviewboard.asterisk.org/r/3898/ ........ Merged
+ revisions 420680 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/tcptls.c, /: tcptls: Avoid compiler warning on non-dev-mode.
+ ........ Merged revisions 420654 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-07 21:37 +0000 [r420435] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Replace sip_tls_read() and
+ resolve the large SDP poll issue. Replace sip_tls_read() and
+ sip_tcp_read() with a single function and resolve the poll/wait
+ issue with large SDP payloads. ASTERISK-18345 #close Reported by:
+ Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835)
+ patch uploaded by Elazar Broad Review:
+ https://reviewboard.asterisk.org/r/3882/ ........ Merged
+ revisions 420434 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-06 16:08 +0000 [r420147] George Joseph <george.joseph at fairview5.com>
+
+ * pbx/pbx_lua.c, main/pbx.c, /: pbx_lua: fix regression with global
+ sym export and context clash by pbx_config. ASTERISK-23818 (lua
+ contexts being overwritten by contexts of the same name in
+ pbx_config) surfaced because pbx_lua, having the
+ AST_MODFLAG_GLOBAL_SYMBOLS set, was always force loaded before
+ pbx_config. Since I couldn't find any reason for pbx_lua to
+ export it's symbols to the rest of Asterisk, I simply changed the
+ flag to AST_MODFLAG_DEFAULT. Problem solved. What I didn't
+ realize was that the symbols need to be exported not because
+ Asterisk needs them but because any external Lua modules like
+ luasql.mysql need the base Lua language APIs exported
+ (ASTERISK-17279). Back to ASTERISK-23818... It looks like there's
+ an issue in pbx.c where context_merge was only merging includes,
+ switches and ignore patterns if the context was already existing
+ AND has extensions, or if the context was brand new. If pbx_lua
+ is loaded before pbx_config, the context will exist BUT pbx_lua,
+ being implemented as a switch, will never place extensions in it,
+ just the switch statement. The result is that when pbx_config
+ loads, it never merges the switch statement created by pbx_lua
+ into the final context. This patch sets pbx_lua's modflag back to
+ AST_MODFLAG_GLOBAL_SYMBOLS and adds an "else if" in context_merge
+ that catches the case where an existing context has includes,
+ switchs or ingore patterns but no actual extensions.
+ ASTERISK-23818 #close Reported by: Dennis Guse Reported by: Timo
+ Teräs Tested by: George Joseph Review:
+ https://reviewboard.asterisk.org/r/3891/ ........ Merged
+ revisions 420146 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-08-05 18:23 +0000 [r420054] Richard Mudgett <rmudgett at digium.com>
+
+ * main/format.c: format.c: Add reason comments for the format_list
+ ordering.
+
+2014-08-04 19:44 +0000 [r419943] Rusty Newton <rnewton at digium.com>
+
+ * main/manager.c, /: Manager - Improve documentation for manager
+ commands Getvar and Setvar. The documentation for these commands
+ did not make it clear that they could accept expressions and
+ functions. Modified to make this clear, but tried not to be
+ overly explicit. ASTERISK-21178 #close Reported by: Rusty Newton
+ Tested by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/3854 ........ Merged revisions
+ 419942 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-28 18:34 +0000 [r419685] Richard Mudgett <rmudgett at digium.com>
+
+ * funcs/func_jitterbuffer.c, apps/app_queue.c,
+ apps/app_speech_utils.c, /, funcs/func_frame_trace.c: datastores:
+ Audit ast_channel_datastore_remove usage. Audit of v1.8 usage of
+ ast_channel_datastore_remove() for datastore memory leaks. *
+ Fixed leaks in app_speech_utils and func_frame_trace. * Fixed
+ app_speech_utils not locking the channel when accessing the
+ channel datastore list. Review:
+ https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of
+ ast_channel_datastore_remove() for datastore memory leaks. *
+ Fixed leak in func_jitterbuffer. Review:
+ https://reviewboard.asterisk.org/r/3860/ ........ Merged
+ revisions 419684 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-25 23:13 +0000 [r419631] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c, /: features.c: Allow appliationmap to use Gosub.
+ Using DYNAMIC_FEATURES with a Gosub application as the mapped
+ application does not work. It does not work because Gosub just
+ pushes the current dialplan context, exten, and priority onto a
+ stack and sets the specified Gosub location. Gosub does not have
+ a dialplan execution loop to run dialplan like Macro. * Made the
+ DYNAMIC_FEATURES application mapping feature call
+ ast_app_exec_macro() and ast_app_exec_sub() for the Macro and
+ Gosub applications respectively. * Backported
+ ast_app_exec_macro() and ast_app_exec_sub() from v11 to execute
+ dialplan routines from the DYNAMIC_FEATURES application mapping
+ feature. NOTE: This issue does not affect v12+ because it already
+ does what this patch implements. AST-1391 #close Reported by:
+ Guenther Kelleter Review:
+ https://reviewboard.asterisk.org/r/3844/ ........ Merged
+ revisions 419630 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-24 17:56 +0000 [r419441] Corey Farrell <git at cfware.com>
+
+ * /, channels/chan_sip.c: chan_sip: sip_subscribe_mwi_destroy
+ should not call sip_destroy sip_subscribe_mwi_destroy calls
+ sip_destroy on the reference counted mwi->call. This results in
+ the fields of mwi->call being freed, but mwi->call itself it
+ leaked. If other code is still using mwi->call it can cause
+ problems. This change uses dialog_unref instead, to balance the
+ ref provided by sip_alloc(). ASTERISK-24087 #close Reported by:
+ Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/
+ ........ Merged revisions 419440 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-24 16:49 +0000 [r419375] Jason Parker <jparker at digium.com>
+
+ * addons/chan_ooh323.c, /: Don't cause Asterisk to exit if
+ ooh323.conf not found. (closes issue ASTERISK-23814) ........
+ Merged revisions 419374 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-23 13:21 +0000 [r419284] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * apps/app_voicemail.c: app_voicemail: use a consistent generator
+ string When updating voicemail.conf when a user changes their
+ pin, change the generator string to be the same as the module
+ name when reading so that the same config_hook will be called.
+ Review: https://reviewboard.asterisk.org/r/3837/
+
+2014-07-22 14:00 +0000 [r419162] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_voicemail_api.c, tests/test_aoc.c,
+ tests/test_astobj2.c, tests/test_config.c,
+ addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c,
+ addons/chan_ooh323.c, tests/test_astobj2_thrash.c, /,
+ apps/app_meetme.c, tests/test_abstract_jb.c, tests/test_logger.c,
+ tests/test_event.c, tests/test_format_api.c,
+ tests/test_hashtab_thrash.c, res/res_jabber.c: Fix more dev-mode
+ build issues ........ Merged revisions 419129 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-15 22:05 +0000 [r418713] Matthew Jordan <mjordan at digium.com>
+
+ * main/manager.c: manager: Return ActionID on nominal responses to
+ PresenceState action When the PresenceState action is executed,
+ the nominal path fails to include the ActionID in the successful
+ response. This patch adds a call to astman_start_ack, which
+ guarantees that an ActionID (if provided) will be sent back to
+ the AMI client. Review: https://reviewboard.asterisk.org/r/3776/
+ ASTERISK-23985 #close
+
+2014-07-15 17:32 +0000 [r418649] Jonathan Rose <jrose at digium.com>
+
+ * funcs/func_uri.c, /: func_uri: URIENCODE/URIDECODE - allow empty
+ strings as argument Previously these two dialplan functions would
+ issue warnings and return failure when an empty string is used as
+ the argument. Now they will not issue a warning and will
+ successfully return an empty string. ASTERISK-23911 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3745/ ........ Merged
+ revisions 418641 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-13 21:51 +0000 [r418465-418505] Corey Farrell <git at cfware.com>
+
+ * /, main/astobj2.c, contrib/scripts/refcounter.py: astobj2: work
+ around REF_DEBUG race which causes out of order log entries *
+ Update refcounter.py to use delta's to track the current
+ reference count. * Use result from internal_ao2_ref to write
+ old_refcount to refs_log. Review:
+ https://reviewboard.asterisk.org/r/3756/ ........ Merged
+ revisions 418504 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_skel.c: Fix minor reference leaks in app_skel and
+ TEST_FRAMEWORK * Cleanup games object in app_skel. * Cleanup
+ stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review:
+ https://reviewboard.asterisk.org/r/3757/
+
+2014-07-11 14:23 +0000 [r418366] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/config.c: config: inform config hook of change when writing
+ file When updated configuration is written back to the conf file
+ - for example when a user changes their voicemail pin, make sure
+ that any config hook that wants to know of changes is informed.
+ Review: https://reviewboard.asterisk.org/r/3708/
+
+2014-07-10 15:35 +0000 [r418323] Matthew Jordan <mjordan at digium.com>
+
+ * include/asterisk/xmpp.h: include/asterisk/xmpp.h: Convert
+ indentation to tabs This is a whitespace only change.
+
+2014-07-10 01:42 +0000 [r418262] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c: chan_dahdi/sig_pri: Fix type mismatch in
+ the idledial feature's channel creation. Square pegs in round
+ holes don't work very well. ........ Merged revisions 418261 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.11.0 Released.
+
+2014-07-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.11.0-rc1 Released.
+
+2014-07-03 21:48 +0000 [r417957] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.h, channels/chan_dahdi.c,
+ configs/chan_dahdi.conf.sample, /, UPGRADE.txt,
+ channels/sig_pri.c: 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/ ........ Merged
+ revisions 417956 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-07-03 11:24 +0000 [r417798] 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) ........ Merged revisions 417797 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-30 19:42 +0000 [r417677] Joshua Colp <jcolp at digium.com>
+
+ * channels/sip/include/sip.h, res/res_rtp_asterisk.c,
+ main/rtp_engine.c, channels/chan_sip.c, UPGRADE.txt,
+ configs/sip.conf.sample, include/asterisk/rtp_engine.h:
+ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS
+ negotiation on RTCP. This change fixes up DTLS support in
+ res_rtp_asterisk so it can accept and provide a SHA-256
+ fingerprint, so it occurs on RTCP, and so it occurs after ICE
+ negotiation completes. Configuration options to chan_sip have
+ also been added to allow behavior to be tweaked (such as forcing
+ the AVP type media transports in SDP). ASTERISK-22961 #close
+ Reported by: Jay Jideliov Review:
+ https://reviewboard.asterisk.org/r/3679/
+
+2014-06-30 03:23 +0000 [r417588] 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.
+ Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916
+ #close Reported by: Alexander Traud patches:
+ sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud
+ (License 6520) ........ Merged revisions 417587 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-27 19:26 +0000 [r417481-417505] 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/ ........ Merged
+ revisions 417500 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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/ ........ Merged
+ revisions 417480 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-06-26 18:25 +0000 [r417310-417419] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_http_websocket.exports.in: res_http_websocket: Export
+ symbol for ast_websocket_set_timeout Thanks to Sean Bright for
+ pointing out that this was missed in #asterisk-dev.
+
+ * 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) ........ Merged revisions 417318
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * UPGRADE.txt, configs/sip.conf.sample, res/res_http_websocket.c,
+ channels/sip/include/sip.h, channels/chan_sip.c,
+ include/asterisk/http_websocket.h: res_http_websocket: Close
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