[asterisk-commits] bebuild: tag 11.14.0-rc1 r427143 - /tags/11.14.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 12:48:46 CST 2014
Author: bebuild
Date: Mon Nov 3 12:48:40 2014
New Revision: 427143
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=427143
Log:
Importing files for 11.14.0-rc1 release.
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tags/11.14.0-rc1/.lastclean (with props)
tags/11.14.0-rc1/.version (with props)
tags/11.14.0-rc1/ChangeLog (with props)
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+2014-11-03 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.14.0-rc1 Released.
+
+2014-11-03 02:31 +0000 [r427019-427087] Corey Farrell <git at cfware.com>
+
+ * apps/app_voicemail.c: Fix compile error caused by review 4138
+ There is no procedure called ast_closeframe, fix code to use
+ ast_closestream. Reported By: Matt Jordan
+
+ * apps/app_voicemail.c, /, 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/
+ ........ Merged revisions 427023 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * funcs/func_jitterbuffer.c, main/abstract_jb.c: func_jitterbuffer:
+ fix frame leaks. Fix code paths where it is possible for frames
+ to leak. Fix uninitialized variable in jb_get_fixed and
+ jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/4128/
+
+2014-10-31 16:40 +0000 [r426927-426931] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * Makefile, /: Fix syntax from commit r426927
+
+ * Makefile, /: install init.d files on GNU/kFreeBSD Review:
+ https://reviewboard.asterisk.org/r/4118/ ........ Merged
+ revisions 426926 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-31 03:25 +0000 [r426860] Matthew Jordan <mjordan at digium.com>
+
+ * channels/sip/include/reqresp_parser.h, /,
+ channels/sip/reqresp_parser.c: 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. ........ Merged revisions 426858 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-31 03:05 +0000 [r426805-426831] Corey Farrell <git at cfware.com>
+
+ * /, contrib/Makefile (added), Makefile: 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/ ........ Merged
+ revisions 426830 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_queue.c: app_queue: fix a couple leaks to struct
+ call_queue in set_member_value set_member_value has a couple
+ leaks to references in the variable q found through testsuite
+ tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES
+ compiler declaration, this is no longer possible with the updated
+ REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/4125/
+
+2014-10-30 09:16 +0000 [r426692] 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/ ........ Merged
+ revisions 426691 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-30 05:56 +0000 [r426666] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Add additional checks for NULL pointers
+ to fix several crashes reported. ASTERISK-24304 #close Reported
+ by: dhanapathy sathya
+
+2014-10-30 01:58 +0000 [r426595-426600] 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) ........ Merged revisions 426599
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged revisions 426594 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-28 20:50 +0000 [r426527] Corey Farrell <git at cfware.com>
+
+ * res/res_fax.c: res_fax: Resolve T38 gateway frame leak. When
+ frames are translated by a fax gateway they need to be freed. The
+ existing call to ast_frfree was unreachable. This change
+ reorganizes fax_gateway_framehook to ensure that ast_frfree is
+ called when needed. ASTERISK-24457 #close Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/4115/
+
+2014-10-28 18:08 +0000 [r426456] mdavenport <mdavenport at localhost>:
+
+ * configs/manager.conf.sample: ASTERISK-23512, correct inaccurate
+ comment in manager.conf.sample
+
+2014-10-28 14:57 +0000 [r426366] Matthew Jordan <mjordan at digium.com>
+
+ * main/manager.c: main/manager: Fix typo in AMI event documentation
+ of "OriginateResponse" The parameter name is "Response", not
+ "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni
+
+2014-10-28 14:55 +0000 [r426291-426359] mdavenport <mdavenport at localhost>:
+
+ * 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
+
+2014-10-28 11:17 +0000 [r426255] Corey Farrell <git at cfware.com>
+
+ * apps/app_queue.c: app_queue: Cleanup ao2_iterator Clean
+ ao2_iterator, resolving reference leak to queue members.
+ ASTERISK-24454 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4111/
+
+2014-10-27 02:45 +0000 [r426141-426209] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_http_websocket.c: res/res_http_websocket: Fix minor nits
+ found by wdoekes on r409681 When Moises committed the fixes for
+ WSS (which was a great patch), wdoekes had a few style nits that
+ were on the review that got missed. This patch resolves what I
+ *think* were all of the ones that were still on the review.
+ Thanks to both moy for the patch, and wdoekes for the reviews.
+ Review: https://reviewboard.asterisk.org/r/3248/
+
+ * 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 ........
+ Merged revisions 426140 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-20 14:10 +0000 [r425986] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, res/res_xmpp.c, res/res_jabber.c, main/tcptls.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)
+
+2014-10-17 13:09 +0000 [r425819] 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) ........ Merged
+ revisions 425818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-16 06:04 +0000 [r425667] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Fix loss of voice after second call
+ drops (on a second line) in case using multiple lines on unistim
+ phones. There is regression was introduced in r391379. Reported
+ by: Rustam Khankishyiev (closes issue ASTERISK-23846)
+
+2014-10-16 01:24 +0000 [r425644] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix a bug where ICE
+ state would get reset when it shouldn't. In the case where the
+ ICE negotiation had not yet started current state would get wiped
+ when it shouldn't. This also removes channel binding as in
+ practice this does not work well with other implementations.
+
+2014-10-15 09:02 +0000 [r425548] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c, /: 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 ........
+ Merged revisions 425547 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-14 16:44 +0000 [r425407-425457] Corey Farrell <git at cfware.com>
+
+ * res/res_fax.c: res_fax: Fix reference leak caused by gateway
+ sessions Fax gateway session objects can be re-used, causing the
+ same gateway session to be added to faxregistry.container more
+ than once. This change causes fax_session_new to remove the
+ reserved session from the container before it's id is changed,
+ ensuring it's possible for the session to be freed.
+ ASTERISK-24392 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4049/
+
+ * /, 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/
+ ........ Merged revisions 425405 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-12 21:08 +0000 [r425360] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Make the ICE transport
+ check case insensitive as some implementations use 'udp'.
+
+2014-10-12 08:13 +0000 [r425287-425297] 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) ........ Merged revisions 425296 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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/ ........ Merged
+ revisions 425291 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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/ ........ Merged
+ revisions 425286 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-10 12:55 +0000 [r425153] 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/
+ ........ Merged revisions 425152 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-10 07:25 +0000 [r425069] 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) ........ Merged revisions 425068 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-09 21:26 +0000 [r425029] Kevin Harwell <kharwell at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Crash if no candidates
+ received for component When starting ice if there is not at least
+ one remote ice candidate with an RTP component asterisk will
+ crash. This is due to an assertion in pjnath as it expects at
+ least one candidate with an RTP component. Added a check to make
+ sure at least one candidate contains an RTP component and at
+ least one candidate has an RTCP component. ASTERISK-24383 #close
+ Review: https://reviewboard.asterisk.org/r/4039/
+
+2014-10-09 08:06 +0000 [r424878] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * 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) ........ Merged
+ revisions 424875 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-08 18:44 +0000 [r424852] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Allow only UDP ICE
+ candidates. The underlying library, pjnath, that res_rtp_asterisk
+ uses for ICE support does not have support for ICE-TCP. As
+ candidates are passed through directly to it this can cause error
+ messages to occur when it receives something unexpected (such as
+ a TCP candidate). This change merely ignores all non-UDP
+ candidates so they never reach pjnath. ASTERISK-24326 #close
+ Reported by: Joshua Colp
+
+2014-10-07 21:30 +0000 [r424787] 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/ ........ Merged
+ revisions 424786 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-10-06 18:36 +0000 [r424690] Matthew Jordan <mjordan at digium.com>
+
+ * main/message.c: message: Don't close an AMI connection on
+ SendMessage action error If SendMessage encounters an error (such
+ as incorrect input provided to the action), it will currently
+ return -1. Actions should only return -1 if the connection to the
+ AMI client should be closed. In this case, SendMessage causing
+ the client to disconnect is inappropriate. This patch causes the
+ action to return 0, which simply causes the action to fail.
+ Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354
+ #close Reported by: Peter Katzmann patches: sendMessage.patch
+ uploaded by Peter Katzmann (License 5968)
+
+2014-10-05 00:41 +0000 [r424550-424578] Corey Farrell <git at cfware.com>
+
+ * main/manager.c: Release AMI connections on shutdown.
+ ASTERISK-24378 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4037/
+
+ * channels/chan_sip.c: chan_sip: Clean leak on error path of
+ process_sdp Resolve leak in process_sdp that occurs in 2 error
+ path's where crypto lines are expected but not provided.
+ ASTERISK-24385 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4045/
+
+ * channels/chan_motif.c: chan_motif: Release format capabilities
+ and config on module load error ASTERISK-24384 #close Reported
+ by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4043/
+
+2014-10-01 10:08 +0000 [r424177-424182] 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) ........ Merged revisions 424181 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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/ ........ Merged
+ revisions 424176 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-30 11:31 +0000 [r424151] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Ensure that the base
+ and mapped address for candidates is present in SDP. This change
+ fixes an issue where ICE candidates put into the SDP did not
+ contain the 'raddr' and 'rport' information for server reflexive
+ and relay candidates. #SIPit31
+
+2014-09-29 21:21 +0000 [r424117] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * contrib/scripts/autosupport: autosupport: Fix bashism. '==' is
+ bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
+ 'case' works better there. Originally committed in r375059 and
+ r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported
+ by: Tzafrir Cohen
+
+2014-09-26 15:18 +0000 [r423983] 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.
+ ........ Merged revisions 423979 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-26 08:23 +0000 [r423916] 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) ........ Merged revisions 423915 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-24 08:49 +0000 [r423801] 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/ ........ Merged
+ revisions 423800 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-22 19:46 +0000 [r423658-423721] 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) ........ Merged revisions 423720 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, main/editline/readline.c: cli.c: Fix tab completion "module
+ load" when MALLOC_DEBUG is enabled. r421600 conflicted with
+ r155763. ASTERISK-24348 #close ........ Merged revisions 423657
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-09-24 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.13.0 Released.
+
+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
[... 31213 lines stripped ...]
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