[asterisk-commits] bebuild: tag 11.15.0-rc1 r429096 - /tags/11.15.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Dec 8 11:06:35 CST 2014
Author: bebuild
Date: Mon Dec 8 11:06:32 2014
New Revision: 429096
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=429096
Log:
Importing files for 11.15.0-rc1 release.
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tags/11.15.0-rc1/.version (with props)
tags/11.15.0-rc1/ChangeLog (with props)
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+2014-12-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.15.0-rc1 Released.
+
+2014-12-06 18:15 +0000 [r429027-429031] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_monitor.c: res/res_monitor: Reset in/out sample counts on
+ Monitor start When repeatedly starting/stopping a Monitor on a
+ channel, the accumulated in/out sample counts are never reset to
+ 0. This can cause inadvertent jumps in the recordings, as the
+ code in the channel core will determine incorrectly that a jump
+ in the recorded file position should occur. Setting the sample
+ counts to 0 simply reflects the initial state a Monitor should be
+ in when it is started, as this is the initial count that would be
+ on the channels at that time. ASTERISK-24573 #close Reported by:
+ Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License
+ 6116)
+
+ * apps/app_meetme.c: apps/app_meetme: Apply default values on
+ initial load with no config file When the app_meetme module is
+ loaded without its configuration file, the module settings aren't
+ initialized. In particular, this impacts the use of logging
+ realtime members. This patch guarantees that we always set the
+ default module settings on initial load. Review:
+ https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close
+ Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno
+ Borges (License 6116)
+
+2014-12-03 16:43 +0000 [r428787-428863] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c: apps/app_voicemail: Fix crash with IMAP
+ when streams are opened simultaneously The UW IMAP library is
+ instrinsically not thread-safe, and relies upon higher level
+ applications to guarantee thread safety. For the most part, this
+ is provided by the vms object, which provides locking for
+ individual streams. Unfortunately, this is not sufficient for
+ calls to mail_open which create the IMAP stream. mail_open can,
+ on some systems, call into a UW IMAP specific function for
+ determining the address of a system based on a hostname,
+ ip_nametoaddr. In the ip6_unix implementation of this function,
+ static variables are used to hold parsing buffers. This can cause
+ a crash if multiple threads attempt to convert a hostname to an
+ address at the same time. Locking on a single mail stream is not
+ sufficient to prevent simultaneous access to these static
+ variables. In the IMAP library, this function can be called from
+ the mail_open and imap_status functions. As the imap_status
+ function is not used by app_voicemail, locking on access to
+ mail_open is sufficient to prevent any mangling of the buffers.
+ Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516
+ #close Reported by: David Duncan Ross Palmer Tested by: David
+ Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David
+ Duncan Ross Palmer (License 6660)
+
+ * pbx/pbx_loopback.c: pbx/pbx_loopback: Speed up switches by
+ avoiding unneeded lookups This patch makes a small rearrangement
+ to only do dialplan lookups during loopback switches if the
+ pattern matches. Prior to this patch, the dialplan lookups were
+ always performed, even when the result would be discarded.
+ Dialplan lookups can be very costly if remote switches - like
+ DUNDi - are present. In those cases extension matching is sped up
+ considerably, making the issue of lost digits more manageable. As
+ collateral damage, 6 trailing spaces were killed. Review:
+ https://reviewboard.asterisk.org/r/4211 ASTERISK-24577 #close
+ Reported by: Birger Harzenetter patches: ast-loopback.patch
+ uploaded by Birger Harzenetter (License 5870)
+
+2014-12-01 13:39 +0000 [r428653] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_record.c: app_record: Fix bug where using the 'k' option
+ and hanging up would trim 1/4 of a second of the recording. The
+ Record dialplan function trims 1/4 of a second from the end of
+ recordings in case they are terminated because of DTMF. When
+ hanging up, however, you don't want this to happen. This change
+ makes it so on hangup this does not occur. ASTERISK-24530 #close
+ Reported by: Ben Smithurst patches: app_record_v2.diff submitted
+ by Ben Smithurst (license 6529) Review:
+ https://reviewboard.asterisk.org/r/4201/
+
+2014-11-21 18:47 +0000 [r428570] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c: manager: Fix could not extend string messages.
+ When shutting down Asterisk that has an active AMI connection,
+ you get several "failed to extend from %d to %d" messages because
+ use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission
+ strings to the event. * Created MAX_AUTH_PERM_STRING to use when
+ creating stack based struct ast_str variables used with the
+ authority_to_str() and user_authority_to_str() functions instead
+ of a variety of magic numbers that could be too small. * Added a
+ special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it
+ will not attempt to add all permission level strings. Review:
+ https://reviewboard.asterisk.org/r/4200/
+
+2014-11-20 16:35 +0000 [r428417] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/acl.c: Fix error with mixed address family ACLs. Prior to
+ this commit, the address family of the first item in an ACL was
+ used to compare all incoming traffic. This could lead to traffic
+ of other IP address families bypassing ACLs. ASTERISK-24469
+ #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff
+ uploaded by Matt Jordan (License #6283) AST-2014-012 ........
+ Merged revisions 428402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-11-20 16:22 +0000 [r428332-428363] Kevin Harwell <kharwell at digium.com>
+
+ * funcs/func_db.c, /: AST-2014-018 - func_db: DB Dialplan function
+ permission escalation via AMI. The DB dialplan function when
+ executed from an external protocol (for instance AMI), could
+ result in a privilege escalation. Asterisk now inhibits the DB
+ function from being executed from an external interface if the
+ live_dangerously option is set to no. ASTERISK-24534 Reported by:
+ Gareth Palmer patches: submitted by Gareth Palmer (license 5169)
+ ........ Merged revisions 428331 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_confbridge.c: AST-2014-017 - app_confbridge: permission
+ escalation/ class authorization. Confbridge dialplan function
+ permission escalation via AMI and inappropriate class
+ authorization on the ConfbridgeStartRecord action. The CONFBRIDGE
+ dialplan function when executed from an external protocol (for
+ instance AMI), could result in a privilege escalation. Also, the
+ AMI action “ConfbridgeStartRecord†could also be used to execute
+ arbitrary system commands without first checking for system
+ access. Asterisk now inhibits the CONFBRIDGE function from being
+ executed from an external interface if the live_dangerously
+ option is set to no. Also, the “ConfbridgeStartRecord†AMI action
+ is now only allowed to execute under a user with system level
+ access. ASTERISK-24490 Reported by: Gareth Palmer
+
+2014-11-20 14:20 +0000 [r428299] Joshua Colp <jcolp at digium.com>
+
+ * main/bridging.c: AST-2014-014: Fix race condition where channels
+ may get stuck in ConfBridge under load. Under load it was
+ possible for the bridging API, and thus ConfBridge, to get
+ channels that may have hung up stuck in it. This is because
+ handling of state transitions for a bridged channel within a
+ bridge was not protected and simply set the new state without
+ regard to the existing state. If the existing state had been hung
+ up this would get overwritten. This change adds locking to
+ protect changing of the state and also takes into consideration
+ the existing state. ASTERISK-24440 #close Reported by: Ben Klang
+ Review: https://reviewboard.asterisk.org/r/4173/
+
+2014-11-19 16:38 +0000 [r428244] Richard Mudgett <rmudgett at digium.com>
+
+ * res/res_calendar.c, channels/chan_sip.c,
+ channels/sip/security_events.c: ast_str: Fix improper member
+ access to struct ast_str members. Accessing members of struct
+ ast_str outside of the string manipulation API routines is
+ invalid since struct ast_str is supposed to be treated as opaque.
+ Review: https://reviewboard.asterisk.org/r/4194/
+
+2014-11-17 15:56 +0000 [r428117] Corey Farrell <git at cfware.com>
+
+ * channels/chan_sip.c: chan_sip: Fix theoretical leak of p->refer.
+ If transmit_refer is called when p->refer is already allocated,
+ it leaks the previous allocation. Updated code to always free
+ previous allocation during a new allocation. Also instead of
+ checking if we have a previous allocation, always create a clean
+ record. ASTERISK-15242 #close Reported by: David Woolley Review:
+ https://reviewboard.asterisk.org/r/4160/
+
+2014-11-17 15:26 +0000 [r428077-428113] Matthew Jordan <mjordan at digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c: apps/app_confbridge:
+ Ensure 'normal' users hear message when last marked leaves When
+ r428077 was made for ASTERISK-24522, it failed to take into
+ account users who are neither wait_marked nor end_marked. These
+ users are *also* supposed to hear the 'leader has left the
+ conference' message. Granted, this behaviour is a bit odd;
+ however, that is how it used to work... and behaviour changes are
+ not good. This patch ensures that if there are any 'normal' users
+ present when the last marked user leaves the conference, the
+ message will still be played to them. Note that this regression
+ was caught by the Asterisk Test Suite's confbridge_nominal test,
+ which has a quirky combination of users.
+
+ * apps/confbridge/conf_state_multi_marked.c: app_confbridge: Don't
+ play leader leaving prompt if no one will hear it Consider the
+ following: - A marked user in a conference - One or more
+ end_marked only users in the conference When the marked users
+ leaves, we will be in the conf_state_multi_marked state. This
+ currently will traverse the users, kicking out any who have the
+ end_marked flags. When they are kicked, a full ast_bridge_remove
+ is immediately called on the channels. At this time, we also
+ unilaterally set the need_prompt flag. When the need_prompt flag
+ is set, we then playback a sound to the bridge informing everyone
+ that the leader has left; however, no one is left in the bridge.
+ This causes some odd behaviour for the end_marked users - they
+ are stuck waiting for the bridge to be unlocked. This results in
+ them waiting for 5 or 6 seconds of dead air before hearing that
+ they've been kicked. Unfortunately, we do have to keep the bridge
+ locked while we're playing back the 'leader-has-left' prompt. If
+ there are any wait_marked users in the conference, this behaviour
+ can't be easily changed - but we do make the case of the
+ end_marked users better with this patch. Review:
+ https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close
+ Reported by: Matt Jordan
+
+2014-11-15 16:51 +0000 [r427952] Matthew Jordan <mjordan at digium.com>
+
+ * cel/cel_odbc.c: cel/cel_odbc: Provide microsecond precision in
+ 'eventtime' column when possible This patch adds microsecond
+ precision when inserting a CEL record into a table with an
+ "eventtime" column of type timestamp, instead of second
+ precision. The documentation (configs/cel_odbc.conf.sample) was
+ already saying that the eventtime column included microseconds
+ precision, but that was not the case. Also, without this patch,
+ if you had a table with an "eventtime" column of type varchar,
+ you had millisecond precision. With this patch, you also get
+ microsecond precision in this case. Review:
+ https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close
+ Reported by: Etienne Lessard patches:
+ cel_odbc_time_precision.patch uploaded by Etienne Lessard
+ (License 6394)
+
+2014-11-14 15:46 +0000 [r427874] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/stun.c: stun: correct attribute string padding to match rfc
+ When sending the USERNAME attribute in an RTP STUN response, the
+ implementation in append_attr_string passed the actual length,
+ instead of padding it up to a multiple of four bytes as required
+ by the RFC 3489. This change adds separate variables for the
+ string and padded attributed lengths, and performs padding
+ correctly. Reported by: Thomas Arimont Review:
+ https://reviewboard.asterisk.org/r/4139/
+
+2014-11-14 14:54 +0000 [r427844] Joshua Colp <jcolp at digium.com>
+
+ * apps/confbridge/conf_state_multi_marked.c: app_confbridge: Play
+ "leader has left" sound even when musiconhold is enabled.
+ Currently if the leader of a conference bridge leaves any
+ participant that has musiconhold enabled will not hear the
+ "leader has left" sound. This is because musiconhold is started
+ and THEN the sound is played. This change makes it so that the
+ sound is played and THEN musiconhold is started. This provides a
+ better experience for users as they may not have known previously
+ why they went back to musiconhold. Review:
+ https://reviewboard.asterisk.org/r/4177/
+
+2014-11-12 16:10 +0000 [r427709] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c: pbx: Fix off-nominal case where a freed extension may
+ still be used. If during the operation of adding an extension a
+ priority is added but fails it is possible for the extension to
+ be freed but still exist in the PBX core. If this occurs
+ subsequent lookups may try to access the extension and end up in
+ freed memory. This change removes the extension from the PBX core
+ when the priority addition fails and then frees the extension.
+ ASTERISK-24444 #close Reported by: Leandro Dardini Review:
+ https://reviewboard.asterisk.org/r/4162/
+
+2014-11-12 13:44 +0000 [r427682] Corey Farrell <git at cfware.com>
+
+ * codecs/ilbc, tests, codecs/speex, apps/confbridge,
+ Makefile.rules: Fix compiler error when using ./configure
+ --enable-dev-mode --enable-coverage When DONT_OPTIMIZE is enabled
+ with dev-mode, it causes a shadow compilation to be done with
+ output to /dev/null. This can cause errors with coverage when GCC
+ attempts to write to /dev/null.gcno. This change disables
+ coverage for the shadow compilation. ASTERISK-24502 #close
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4151/
+
+2014-11-09 07:56 +0000 [r427641] Corey Farrell <git at cfware.com>
+
+ * main/manager.c: manager: Fix HTTP connection reference leaks. Fix
+ reference leak that happens if (session && !blastaway).
+ ASTERISK-24505 #close Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/4153/
+
+2014-11-09 00:59 +0000 [r427607-427617] Matthew Jordan <mjordan at digium.com>
+
+ * configs/features.conf.sample: configs/features.conf: Add
+ documentation noting potential chan_agent conflict In chan_agent,
+ a '*' is used by default to terminate a bridge with a caller.
+ This can lead to all sorts of problems if '*' is used by a
+ feature in features.conf, as the chan_agent disconnect '*' may be
+ detected first. This patch adds a documentation snippet to
+ features.conf so that users who attempt to use features with
+ agents know of the potential conflict. ASTERISK-20402 #close
+ Reported by: Matt Riddell patches: features.conf.diff uploaded by
+ Matt Riddell (License 5023)
+
+ * channels/chan_mgcp.c: channels/chan_mgcp: Fix regression which
+ causes gateways to be skipped In r227276, a while loop was turned
+ into a for loop. Unfortunately, a portion of the while loop was
+ left in the code such that, when a static gateway is encountered
+ in the list of MGCP gateways, the next gateway would be skipped.
+ At best, we would simply flip past a gateway; at worst, this
+ could lead to a crash. ASTERISK-24500 #close Reported by: Xavier
+ Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne
+ (License 6657)
+
+ * addons/chan_mobile.c: addons/chan_mobile: Increase buffer size of
+ UCS2 encoded SMS messages When UCS2 character encoding is used,
+ one symbol in national language can be expanded to 4 bytes. The
+ current buffer used for receiving message in do_monitor_phone is
+ 256 bytes, which is not large enough for incoming messages. For
+ example: * AT+CMGR phone response prefix '+CMGR: "REC
+ UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes *
+ SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone
+ response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null
+ character - 1 byte This results in a needed buffer size of 349
+ bytes. Hence, this patch opts for a 350 byte buffer.
+ ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches:
+ chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651)
+ chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651)
+
+2014-11-08 17:28 +0000 [r427554] Corey Farrell <git at cfware.com>
+
+ * channels/chan_console.c: chan_console: Fix reference leaks to
+ pvt. Fix a bunch of calls to get_active_pvt where the reference
+ is never released. ASTERISK-24504 #close Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/4152/
+
+2014-11-06 12:10 +0000 [r427381-427464] Corey Farrell <git at cfware.com>
+
+ * main/file.c: main/file.c: fix possible extra ast_module_unref to
+ format modules. fn_wrapper only adds a reference to the format's
+ module if the file was able to be opened. If not this causes an
+ unmatched ast_module_unref in filestream_destructor. Move
+ ast_module_ref to get_stream. ASTERISK-24492 #close Reported by:
+ Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/
+
+ * include/asterisk/stringfields.h, /, main/utils.c: Fix unintential
+ memory retention in stringfields. * Fix missing / unreachable
+ calls to __ast_string_field_release_active. * Reset pool->used to
+ zero when the current pool->active reaches zero. ASTERISK-24307
+ #close Reported by: Etienne Lessard Tested by: ibercom, Etienne
+ Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........
+ Merged revisions 427380 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-11-06 02:26 +0000 [r427328-427354] George Joseph <george.joseph at fairview5.com>
+
+ * tests/test_strings.c: test_strings: Remove string tests that
+ exercise asserts. Since unit tests are run with DO_CRASH, those
+ tests were causing the test to fail. Tested-by: George Joseph
+
+ * main/config.c, tests/test_strings.c, include/asterisk/test.h,
+ include/asterisk/utils.h, main/utils.c, pbx/pbx_config.c: config:
+ Make text_file_save and 'dialplan save' escape semicolons in
+ values. When a config file is read, an unescaped semicolon
+ signals comments which are stripped from the value before it's
+ stored. Escaped semicolons are then unescaped and become part of
+ the value. Both of these behaviors are normal and expected. When
+ the config is serialized either by 'dialplan save' or
+ AMI/UpdateConfig however, the now unescaped semicolons are
+ written as-is. If you actually reload the file just saved, the
+ unescaped semicolons are now treated as start of comments. Since
+ true comments are stripped on read, any semicolons in
+ ast_variable.value must have been escaped originally. This patch
+ re-escapes semicolons in ast_variable.values before they're
+ written to file either by 'dialplan save' or
+ config/ast_config_text_file_save which is called by
+ AMI/UpdateConfig. I also fixed a few pre-existing formatting
+ issues nearby in pbx_config.c Tested-by: George Joseph
+ ASTERISK-20127 #close Review:
+ https://reviewboard.asterisk.org/r/4132/
+
+2014-11-10 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.14.0 Released.
+
+2014-11-07 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.14.0-rc2 Released.
+
+2014-11-06 09:05 +0000 [r427381] Corey Farrell <git at cfware.com>
+
+ * Fix unintential memory retention in stringfields.
+
+ * Fix missing / unreachable calls to
+ __ast_string_field_release_active.
+ * Reset pool->used to zero when the current pool->active reaches
+ zero.
+
+ ASTERISK-24307 #close
+ Reported by: Etienne Lessard
+ Tested by: ibercom, Etienne Lessard
+ Review: https://reviewboard.asterisk.org/r/4114/
+
+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>
+
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