[asterisk-commits] bebuild: tag 1.8.32.0-rc1 r427139 - /tags/1.8.32.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 12:42:11 CST 2014
Author: bebuild
Date: Mon Nov 3 12:42:08 2014
New Revision: 427139
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=427139
Log:
Importing files for 1.8.32.0-rc1 release.
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tags/1.8.32.0-rc1/.version (with props)
tags/1.8.32.0-rc1/ChangeLog (with props)
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--- tags/1.8.32.0-rc1/ChangeLog (added)
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+2014-11-03 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.32.0-rc1 Released.
+
+2014-11-02 08:01 +0000 [r427023] Corey Farrell <git at cfware.com>
+
+ * main/app.c: Fix ast_writestream leaks Fix cleanup in
+ __ast_play_and_record where others[x] may be leaked. This was
+ caught where prepend != NULL && outmsg != NULL, once realfile[x]
+ == NULL any further others[x] would be leaked. A cleanup block
+ was also added for prepend != NULL && outmsg == NULL. 11+: Fix
+ leak of ast_writestream recording_fs in
+ app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by:
+ Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/
+
+2014-10-31 16:37 +0000 [r426926-426929] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * Makefile: Fix syntax from r426926
+
+ * Makefile: install init.d files on GNU/kFreeBSD Review:
+ https://reviewboard.asterisk.org/r/4118/
+
+2014-10-31 03:24 +0000 [r426858] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/reqresp_parser.c,
+ channels/sip/include/reqresp_parser.h:
+ channels/sip/reqresp_parser: Fix unit tests for r426594 When
+ r426594 was made, it did not take into account a unit test that
+ verified that the function properly populated the unsupported
+ buffer. The function would previously memset the buffer if it
+ detected it had any contents; since this function can now be
+ called iteratively on successive headers, the unit tests would
+ now fail. This patch updates the unit tests to reset the buffer
+ themselves between successive calls, and updates the
+ documentation of the function to note that this is now required.
+
+2014-10-31 03:03 +0000 [r426830] Corey Farrell <git at cfware.com>
+
+ * Makefile, contrib/Makefile (added): REF_DEBUG: Install
+ refcounter.py to $(ASTDATADIR)/scripts This change ensures
+ refcounter.py is installed to a place where it can be found by
+ the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432
+ #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4094/
+
+2014-10-30 09:11 +0000 [r426691] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_voicemail.c: app_voicemail: Fix unchecked bounds of
+ myArray in IMAP_STORAGE. In update_messages_by_imapuser(),
+ messages were appended to a finite array which resulted in a
+ crash when an IMAP mailbox contained more than 256 entries. This
+ memory is now dynamically increased as needed. Observe that this
+ patch adds a bunch of XXX's to questionable code. See the review
+ (url below) for more information. ASTERISK-24190 #close Reported
+ by: Nick Adams Tested by: Nick Adams Review:
+ https://reviewboard.asterisk.org/r/4126/
+
+2014-10-30 01:57 +0000 [r426594-426599] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: channels/chan_sip: Add improved support for
+ 4xx error codes This patch adds support for 414, 493, 479, and a
+ stray 400 response in REGISTER response handling. This helps
+ interoperability in a number of scenarios. Review:
+ https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch
+ uploaded by oej (License 5267)
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c:
+ channels/chan_sip: Support mutltiple Supported and Required
+ headers A SIP request may contain multiple Supported: and
+ Required: headers. Currently, chan_sip only parses the first
+ Supported/Required header it finds. This patch adds support for
+ multiple Supported/Required headers for INVITE requests. Review:
+ https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close
+ Reported by: Olle Johansson patches: rb2478.patch uploaded by oej
+ (License 5267)
+
+2014-10-28 18:09 +0000 [r426295-426460] mdavenport <mdavenport at localhost>:
+
+ * configs/manager.conf.sample: ASTERISK-23512, correct inaccurate
+ comment in manager.conf.sample
+
+ * res/res_agi.c: ASTERISK-24323, fix bug in documentation of AGI
+ STREAM FILE CONTROL
+
+ * configs/extensions.conf.sample: ASTERISK-24419, fix incorrect
+ syntax for setting language in extensions.conf.sample
+
+2014-10-27 01:39 +0000 [r426140] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_srtp.c: res/res_srtp: Fix include issue for libsrtp 1.5.0
+ In libsrtp 1.5.0, crypto_get_random is no longer resolved simply
+ by including srtp.h. Now, one must include crypto_kernel.h as
+ well. As it turns out, this header file has been provided by the
+ library since 2006, so this is a relatively benign change.
+ ASTERISK-24436 #close Reported by: Patrick Laimbock
+
+2014-10-20 14:09 +0000 [r425985] Matthew Jordan <mjordan at digium.com>
+
+ * main/tcptls.c, UPGRADE.txt, res/res_jabber.c: AST-2014-011: Fix
+ POODLE security issues There are two aspects to the
+ vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch
+ updates the module to use TLSv1+. At this time, it does not
+ refactor res_jabber/res_xmpp to use the TCP/TLS core, which
+ should be done as an improvement at a latter date. (2) The
+ TCP/TLS core, when tlsclientmethod/sslclientmethod is left
+ unspecified, will default to the OpenSSL SSLv23_method. This
+ method allows for all encryption methods, including SSLv2/SSLv3.
+ A MITM can exploit this by forcing a fallback to SSLv3, which
+ leaves the server vulnerable to POODLE. This patch adds WARNINGS
+ if a user uses SSLv2/SSLv3 in their configuration, and explicitly
+ disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients,
+ Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is
+ explicitly chosen. For TLS servers, Asterisk will no longer
+ support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the
+ vulnerability and providing a patch for the res_jabber/res_xmpp
+ modules. Review: https://reviewboard.asterisk.org/r/4096/
+ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck,
+ opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch
+ uploaded by abelbeck (License 5903)
+ asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License
+ 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283)
+ AST-2014-011-11.diff uploaded by mjordan (License 6283)
+ AST-2014-011-12.diff uploaded by mjordan (License 6283)
+
+2014-10-17 13:07 +0000 [r425818] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: channels/chan_sip: Respect outboundproxy
+ setting when sending qualify requests The outboundproxy setting
+ is currently ignored when sending OPTIONS requests as a result of
+ the qualify setting. This means that if an Asterisk server is
+ unable to send the packet directly to a peer, it is unable to
+ qualify any non-inbound registered peer (e.g. a peer SIP Trunk).
+ This patch grabs the outboundproxy information for a peer when a
+ qualify attempt is being constructed and, if it finds the
+ information, uses it when sending the OPTIONS request. Review:
+ https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close
+ Reported by: Damian Ivereigh patches: outboundproxy-dai.patch
+ uploaded by Damian Ivereigh (License 6632)
+
+2014-10-15 08:39 +0000 [r425547] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: : Resolve module reference leak caused by
+ reserved sessions chan_ooh323: fix rtptimeout general value
+ checking correct condition to check rtptimeout in [general]
+ config section ASTERISK-24393 #close Reported by: Dmitry Melekhov
+ Tested by: Dmitry Melekhov Patches: ASTERISK-24393.patch
+
+2014-10-14 16:16 +0000 [r425405] Corey Farrell <git at cfware.com>
+
+ * res/res_fax.c: res_fax: Resolve module reference leak caused by
+ reserved sessions Remove reference to module providing reserved
+ session after adding a reference to the final module. This
+ re-reference is done to ensure that module references are correct
+ even if the final session selects a different module than the
+ reserved session. ASTERISK-18923 #close Reported by: Grigoriy
+ Puzankin Review: https://reviewboard.asterisk.org/r/4048/
+
+2014-10-12 08:10 +0000 [r425286-425296] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip: Fix so asterisk won't send
+ reINVITE after a BYE. After a reINVITE glare situation, Asterisk
+ would re-send the reINVITE even though the call had been hung up
+ in the mean time. This patch unschedules the reinvite when
+ handling the BYE. ASTERISK-22791 #close Reported by: Paolo
+ Compagnini Tested by: Paolo Compagnini Review:
+ https://reviewboard.asterisk.org/r/4056/ (testcase is in review
+ r4055)
+
+ * Makefile: build: Relax badshell tilde test to allow for ~ in
+ middle of DESTDIR. The main Makefile has a target test called
+ 'badshell' that tests if DESTDIR does not happen to have an
+ an-expanded tilde (~). This might be the case if you run: make
+ install DESTDIR=~/somewhere/ That test also disallowed valid
+ tildes in directory names. The test is now changed to only
+ trigger on a tilde at the start of the path. ASTERISK-13797
+ #close Reported by: Tzafrir Cohen Review:
+ https://reviewboard.asterisk.org/r/4064/
+
+ * res/res_calendar_ews.c: res_calendar_ews: Relax neon version
+ check to work with 0.30 too. Allow res_calendar_ews to work not
+ only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close
+ Reported by: Tzafrir Cohen Review:
+ https://reviewboard.asterisk.org/r/4068/
+
+2014-10-10 12:55 +0000 [r425152] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_callerid.c, main/callerid.c: CallerID: Fix parsing
+ regression This fixes a regression in callerid parsing introduced
+ when another bug was fixed. This bug occurred when the name was
+ composed entirely of DTMF keys and quoted without a number
+ section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard
+ Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by
+ Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/
+
+2014-10-10 07:24 +0000 [r424875-425068] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip: Fix dialog leak resulting from
+ missing ACK to re-INVITE. If a device re-INVITEs at the same time
+ as the dialog is hung up, and if then the ACK to the re-INVITE
+ never reaches Asterisk, chan_sip would fail to destroy the dialog
+ after a while. This resulted in (most prominently) file handle
+ leaks. (Patch reindented by me.) ASTERISK-20784 #close
+ ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal
+ Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle
+ (License #5334) patch_asterisk_20784.txt uploaded by Nitesh
+ Bansal (License #6418) Reviewboard:
+ https://reviewboard.asterisk.org/r/4052/ (testcase can be found
+ at r4051)
+
+ * channels/chan_sip.c: Ouch! Accidental commit of wrong file in
+ 424875.
+
+ * channels/chan_sip.c, contrib/scripts/safe_asterisk:
+ safe_asterisk: Don't automatically exceed MAXFILES value of 2^20.
+ On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max
+ divided by two can exceed the per-process file limit of 2^20.
+ This patch ensures the value is capped. (Patch cleaned up by me.)
+ ASTERISK-24011 #close Reported by: Michael Myles Patches:
+ safe_asterisk-ulimit.diff uploaded by Michael Myles (License
+ #6626)
+
+2014-10-07 21:28 +0000 [r424786] Corey Farrell <git at cfware.com>
+
+ * main/astobj2.c: astobj2: Correct REF_DEBUG false leak report When
+ ao2_callback is run with OBJ_MULTIPLE and not OBJ_NODATA it
+ allocates a temporary container in a way that does not record
+ REF_DEBUG log entries. This changes that container to correctly
+ record unref's when the container is freed. ASTERISK-24390 #close
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4047/
+
+2014-10-01 10:06 +0000 [r424176-424181] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip: Simplify some unref code by
+ removing unlink_peer_from_tables. ASTERISK-22945 #related
+ Reported by: ibercom Patches:
+ asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License
+ #6599)
+
+ * channels/chan_sip.c: chan_sip: Remove excess ref of realtime peer
+ before sip_poke_peer. The peer is referenced at the end of
+ sip_poke_peer, it should not get an extra ref before the call to
+ sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close
+ Reported by: ibercom Tested by: Yuriy Gorlichenko Patches:
+ asterisk11.patch uploaded by ibercom (License #6599) Review:
+ https://reviewboard.asterisk.org/r/4031/
+
+2014-09-26 15:16 +0000 [r423979] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_fax.c: res_fax: Fix out of bounds error in
+ update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy
+ Laine Patches: res_fax_bounds.patch (license #6561) patch
+ uploaded by Jeremy Laine Modified patch to not use magic numbers.
+
+2014-09-26 08:22 +0000 [r423915] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * doc/asterisk.8: docs: Escape unescaped minus sign in asterisk.8
+ manpage. ASTERISK-23768 #close Reported by: Jeremy Lainé Patches:
+ escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License
+ #6561)
+
+2014-09-24 08:48 +0000 [r423800] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip: Unref outbound proxy structure on
+ dialog/pvt destruction. Make sure outbound proxy refs are always
+ unreffed on dialog destruction. Review:
+ https://reviewboard.asterisk.org/r/4016/
+
+2014-09-22 19:45 +0000 [r423657-423720] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip: On INVITE retransmission, don't
+ add an extra 503 response. INVITE arrives to asterisk, asterisk
+ responds Busy(). If the INVITE is retransmitted, asterisk would
+ generate a 503 in addition to the 486. Thanks Torrey Searle for
+ providing a working regression test. ASTERISK-24335 #close
+ Review: https://reviewboard.asterisk.org/r/4003/ Patches:
+ retrans_486_invite.patch uploaded by Torrey Searle (License
+ #5334)
+
+ * main/editline/readline.c: cli.c: Fix tab completion "module load"
+ when MALLOC_DEBUG is enabled. r421600 conflicted with r155763.
+ ASTERISK-24348 #close
+
+2014-09-24 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.31.0 Released.
+
+2014-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.31.0-rc1 Released.
+
+2014-09-18 16:08 +0000 [r423349] 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/
+
+2014-09-18 14:37 +0000 [r423276] 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/
+
+2014-09-14 15:48 +0000 [r423066] 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/
+
+2014-09-12 18:17 +0000 [r423006] 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/
+
+2014-09-10 15:58 +0000 [r422900] 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/
+
+2014-09-07 00:07 +0000 [r422789] Rusty Newton <rnewton at digium.com>
+
+ * sounds/Makefile, sounds/sounds.xml: 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
+
+2014-09-04 19:51 +0000 [r422584] 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/
+
+2014-08-30 17:19 +0000 [r422439] 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/
+
+2014-08-29 19:38 +0000 [r422293-422376] 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)
+
+ * 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)
+
+ * 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.
+
+2014-08-27 14:25 +0000 [r422112] Kinsey Moore <kmoore at digium.com>
+
+ * include/asterisk/utils.h, channels/chan_sip.c,
+ tests/test_callerid.c (added), tests/test_utils.c,
+ main/callerid.c, main/utils.c: 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/
+
+2014-08-25 16:00 +0000 [r421976] 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/
+
+2014-08-21 22:01 +0000 [r421799] 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.
+
+2014-08-21 17:32 +0000 [r421717] 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)
+
+2014-08-20 22:13 +0000 [r421600] 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().
+
+2014-08-19 19:38 +0000 [r421442] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c: AMI Docs: Fix Status channel parameter
+ optionality
+
+2014-08-18 20:14 +0000 [r421327] 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/
+
+2014-08-17 23:06 +0000 [r421227-421232] 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)
+
+ * 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)
+
+2014-08-15 14:43 +0000 [r421059-421125] 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.
+
+ * main/file.c: main/file: Move test event to emit PLAYBACK event
+ more consistently This is being done in advance of the test for
+ ASTERISK-23953
+
+2014-08-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.30.0 Released.
+
+2014-08-11 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.30.0-rc1 Released.
+
+2014-08-11 10:24 +0000 [r420654-420680] 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/
+
+ * main/tcptls.c: tcptls: Avoid compiler warning on non-dev-mode.
+
+2014-08-07 21:25 +0000 [r420434] 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/
+
+2014-08-06 16:05 +0000 [r420146] 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/
+
+2014-08-04 19:42 +0000 [r419942] 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
+
+2014-07-28 18:27 +0000 [r419684] Richard Mudgett <rmudgett at digium.com>
+
+ * 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/
+
+2014-07-25 23:04 +0000 [r419630] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/app.h, main/features.c, main/app.c,
+ apps/app_stack.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/
+
+2014-07-24 17:55 +0000 [r419440] 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/
+
+2014-07-24 16:47 +0000 [r419374] 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)
+
+2014-07-22 13:17 +0000 [r419129] Kinsey Moore <kmoore at digium.com>
+
+ * addons/chan_ooh323.c, tests/test_astobj2_thrash.c,
+ apps/app_meetme.c, tests/test_logger.c,
+ addons/ooh323c/src/decode.c, tests/test_event.c,
+ tests/test_aoc.c, tests/test_hashtab_thrash.c,
+ tests/test_astobj2.c, addons/ooh323c/src/printHandler.c,
+ addons/ooh323c/src/ooq931.c: Fix more dev-mode build issues
+
+2014-07-15 17:19 +0000 [r418641] 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/
+
+2014-07-13 21:47 +0000 [r418504] 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/
+
+2014-07-10 01:23 +0000 [r418261] 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.
+
+2014-07-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.29.0 Released.
+
+2014-07-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.29.0-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
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