[svn-commits] bebuild: tag 1.8.25.0-rc1 r402102 - /tags/1.8.25.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Oct 28 08:56:10 CDT 2013
Author: bebuild
Date: Mon Oct 28 08:56:08 2013
New Revision: 402102
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402102
Log:
Importing files for 1.8.25.0-rc1 release.
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tags/1.8.25.0-rc1/ChangeLog (with props)
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+2013-10-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.25.0-rc1 Released.
+
+2013-10-25 21:51 +0000 [r401959-402000] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * include/asterisk/rtp_engine.h, main/rtp_engine.c: rtp_engine: fix
+ rtp payloads copy and improve argument names In function
+ ast_rtp_instance_early _bridge_make_compatible the use of
+ instance 0/1 as arguments doesn't clearly communicate a direction
+ that the copying of payloads from the source channel to the
+ destination channel will occur, making it more probable to have
+ the arguments to ast_rtp_codecs_payloads_copy() put in the
+ reverse order. This patch renames the arguments with _dst and
+ _src suffixes and corrects the copy direction.
+
+ * include/asterisk/pbx.h, main/pbx.c: pbx.c: fix confused match
+ caller id that deleted exten still in hash This fixes a bug where
+ a zero length callerid match adjacent to a no match callerid
+ extension entry would be deleted together, which then resulted in
+ hashtable references to free'd memory. A third state of the
+ matchcid value has been added to indicate match to any extension
+ which allows enforcing comparison of matchcid on/off without
+ errors. (closes issue AST-1235) Reported by: Guenther Kelleter
+ Review: https://reviewboard.asterisk.org/r/2930/
+
+2013-10-25 17:21 +0000 [r401619-401914] Jonathan Rose <jrose at digium.com>
+
+ * utils/clicompat.c: Put clicompat-r2.patch back in We've figured
+ out how to resolve the problems this was causing in 12/trunk, so
+ this can go back in now. (issue ASTERISK-22467) Reported by:
+ Corey Farrell Patches: clicompat-r2.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * utils/clicompat.c: revert clicompat-r2.patch from r401704 Patch
+ caused the following build errors against testsuite
+ https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244
+ (issue ASTERISK-22467) Reported by: Corey Farrell
+
+ * main/utils.c: utils: Fix memory leaks and missed unregistration
+ of CLI commands on shutdown Final set of patches in a series of
+ memory leak/cleanup patches by Corey Farrell (closes issue
+ ASTERISK-22467) Reported by: Corey Farrell Patches:
+ main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
+ main-utils-11.patch uploaded by coreyfarrell (license 5909)
+ main-utils-12up.patch uploaded by coreyfarrell (license 5909)
+
+ * tests/test_linkedlists.c: test_linkedlists: Fix memory leak
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ test_linkedlists-1.8.patch uploaded by coreyfarrell (license
+ 5909) test_linkedlists-11up.patch uploaded by coreyfarrell
+ (license 5909)
+
+ * main/jitterbuf.c: jitterbuf: Fix memory leak on jitter buffer
+ reset (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ jitterbuf-jb_reset-leak-1.8.patch
+ jitterbuf-jb_reset-leak-11up.patch
+
+ * main/astobj2.c: astobj2: Unregister debug CLI commands at exit
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell
+ (license 5909) astobj2-clean-debug-cli-12up.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * apps/app_voicemail.c: app_voicemail: Memory Leaks against tests
+ (issue ASTERISK-22467) Reported by: Corey Farrell Patches:
+ app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
+ app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
+
+ * main/asterisk.c, utils/clicompat.c, channels/chan_dahdi.c,
+ codecs/ilbc/doCPLC.c, main/data.c, main/app.c: memory leaks:
+ Memory leak cleanup patch by Corey Farrell (second set) Also
+ covers ast_app_parse_timelen-fail-zero-length.patch, but the
+ patch was replaced with one of my own. (issue ASTERISK-22467)
+ Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch
+ uploaded by coreyfarrell (license 5909) clicompat-r2.patch
+ uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch
+ uploaded by coreyfarrell (license 5909)
+ data-cleanup-test-registration.patch uploaded by coreyfarrell
+ (license 5909) main-asterisk-kill-listener.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * tests/test_dlinklists.c, funcs/func_math.c,
+ channels/sip/reqresp_parser.c, main/test.c,
+ main/editline/readline.c: memory leaks: Memory leak cleanup patch
+ by Corey Farrell (first set) (issue ASTERSIK-22467) Reported by:
+ Corey Farrell Patches:
+ chan_sip-parse_contact_header_test-free-contacts.patch uploaded
+ by coreyfarrell (license 5909) cli-filename-completion-leak.patch
+ uploaded by coreyfarrell (license 5909) func_math.patch uploaded
+ by corefarrell (license 5909) main-test-cleanup.patch uploaded by
+ coreyfarrell (license 5909) test_dlinklists.patch uploaded by
+ coreyfarrell (license 5909)
+
+ * main/translate.c, res/res_rtp_asterisk.c: res_rtp_asterisk:
+ Address jittery DTMF events in RTP streams (closes issue
+ ASTERISK-21170) Reported by: NITESH BANSAL Patches:
+ dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
+ Review: https://reviewboard.asterisk.org/r/2938/
+
+2013-10-23 16:34 +0000 [r401577] Richard Mudgett <rmudgett at digium.com>
+
+ * cdr/cdr_adaptive_odbc.c: cdr_adaptive_odbc: Also apply a filter
+ when the CDR value is empty. Extra CDR records are written if a
+ filtered CDR value is empty because the filter is not checked.
+ (closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria
+
+2013-10-23 15:19 +0000 [r401537] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_mgcp.c: chan_mgcp: Properly handle malformed media
+ lines This corrects a situation in which a media line was not
+ parsed properly and resulted in a crash. (closes issue
+ ASTERISK-21190) Reported by: adomjan Patches:
+ chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
+
+2013-10-23 11:10 +0000 [r401497] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix an issue where an incompatible audio
+ format may be added to SDP. If preferred codecs included any
+ non-audio format the code would mistakenly add the audio format,
+ even if it was not a joint capability with the remote side.
+ (closes issue ASTERISK-21131) Reported by: nbougues Patches:
+ patch_unsupported_codec_1.8.patch uploaded by nbougues (license
+ 6470)
+
+2013-10-22 22:36 +0000 [r401445] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix crash when RTCP is
+ not available during SSRC change In r400089, a patch was put in
+ to correct erroneous RTCP statistic resets. Unfortunately,
+ ast_rtp_read can be called on an RTP instance that does not have
+ RTCP information. This patch prevents that crash by only
+ resetting the statistics if we do actually have an RTCP instance.
+ (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John
+ Bigelow
+
+2013-10-22 00:13 +0000 [r401378] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_analog.c: chan_dahdi: Fix unable to get index
+ warning when transferring an analog call. Transferring an analog
+ call using flashhooks generated an unable to get index WARNING
+ message when the transfer is completed. * Removed unnecessary
+ analog subchannel shell games when transferring a call using
+ flashhooks. Thanks to Tzafrir Cohen for mentioning this in a
+ comment on issue ASTERISK-22720.
+
+2013-10-21 19:45 +0000 [r401325] Kevin Harwell <kharwell at digium.com>
+
+ * main/editline/term.c: Segfault in LIBEDIT_INTERNAL after
+ tgetstr(), when libncurses5-dev isn't installed Include the
+ appropriate declarations when not using termcap, but term+curses
+ and [n]curses do not exist. (closes issue ASTERISK-22351)
+ Reported by: A. Iglesias Patches:
+ issueA22351_libedit_internal_without_ncurses_dev.patch uploaded
+ by wdoekes (license 5674)
+
+2013-10-18 14:40 +0000 [r401178] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * main/channel.c: Properly copy/remove the device state cache flag
+ over a masquerade. In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE
+ flag was added that tells the devstate system to not cache states
+ for non-real devices. However, when optimizing away channels
+ (ast_do_masquerade), that flag wasn't copied. In my case, using
+ Local devices as queue members created a situation where the
+ endpoint was considered in use, but the state change of the
+ device being available again was ignored (not cached). The
+ endpoint channel was optimized into the (previously) Local
+ channel, but kept the do-not-cache flag. The end result being
+ that the queue member apparently stayed in use forever. (closes
+ issue ASTERISK-22718) Reported by: Walter Doekes Review:
+ https://reviewboard.asterisk.org/r/2925/
+
+2013-10-17 15:22 +0000 [r401119] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_jabber.c: Reduce log level of a non-pubsub error message
+ Drop an error log message to debug level 1 since distributed
+ device state functions correctly when receiving this message and
+ it spams the logs. (closes issue ASTERISK-22410) Reported by:
+ abelbeck Patches:
+ asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch
+ uploaded by abelbeck (License 5903)
+
+2013-10-16 11:04 +0000 [r401049] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_queue.c: Don't check all realtime queues when doing
+ "queue show some_queue". When using realtime queues, queues have
+ to be fetched from the database every now and then to see if any
+ info has been changed or to see if the queue has been removed.
+ When fetching info for an individual queue, the pruning of other
+ queues is unnecessarily costly. Review:
+ https://reviewboard.asterisk.org/r/2907/
+
+2013-10-15 14:52 +0000 [r400970] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Prevent chan_sip from sending duplicate
+ BYEs. When a 200 OK for an initial INVITE is received, we were
+ doing the right thing by ACKing and sending an immediate BYE.
+ However, we also were doing the wrong thing and queuing an answer
+ frame, thus causing the call to be answered. This would cause the
+ call to be hung up by the channel thread, thus resulting in a
+ second BYE being sent out. In this fix, I also have set the
+ hangupcause to be correct since the initial BYE being sent by
+ Asterisk had an unknown hangup cause. I have changed to using
+ "Bearer capabilty not available" since the call was hung up due
+ to an SDP offer/answer error. (closes issue ASTERISK-22621)
+ reported by Kinsey Moore
+
+2013-10-14 21:40 +0000 [r400907] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: Reflect the set software gain
+ in the CLI "dahdi show channel" output. * Remember the swgain
+ setting from CLI "dahdi set swgain" command so the CLI "dahdi
+ show channel" output will reflect the current setting. * Updated
+ CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
+ (issue ASTERISK-22429) Reported by: Jaco Kroon Patches:
+ jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded
+ by rmudgett
+
+2013-10-14 21:32 +0000 [r400906] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Do not increment the SDP version between 183
+ and 200 responses. Bumping the SDP version number can cause
+ interoperability problems since receivers of the responses will
+ expect that a 200 SDP will be identical to a previous 183 SDP.
+ (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches:
+ dont-increment-session-version-in-2xx-after-183.patch uploaded by
+ NITESH BANSAL (License #6418)
+
+2013-10-08 22:26 +0000 [r400694-400767] Kinsey Moore <kmoore at digium.com>
+
+ * configure, configure.ac: Add warning when compiling with iODBC
+ support When running configure, libiodbc2 development headers
+ will fulfill the requirement for ODBC development headers, but
+ will not function properly. This adds a warning when libiodbc2
+ development headers are detected instead of unixodbc development
+ headers. (closes issue ASTERISK-22459) Reported by: Patrick
+ Maille Tested by: Walter Doekes Patches:
+ issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes
+ (License 5674)
+
+ * funcs/func_config.c: Fix func_config list entry allocation The
+ AST_CONFIG dialplan function defined in func_config.c allocates
+ its config file list entries using ast_malloc. List entry
+ allocations destined for use with Asterisk's linked list API must
+ be ast_calloc()d or otherwise initialized so that list pointers
+ are set to NULL. These uses of ast_malloc have been replaced by
+ ast_calloc to prevent dereferencing of uninitialized pointer
+ values when traversing the list. (closes issue ASTERISK-22483)
+ Reported by: Brian Scott
+
+2013-10-06 17:07 +0000 [r400622] Michael L. Young <elgueromexicano at gmail.com>
+
+ * apps/app_queue.c: Fix Regression With Queuelog EXITWITHKEY Only
+ Logging Two Out Of Four Fields Commit r62462 added two extra
+ fields for logging "the original position the caller entered the
+ queue at, and the amount of time the caller was waiting in the
+ queue." But when r75969 was merged from 1.4 into trunk (r75977),
+ these two fields disappeared. Those two extra fields were not
+ logged in 1.4 and when the patch was merged, those fields went
+ away. Therefore, this is a regression and was caught by the
+ reporter because he was reading the awesome "Asterisk: The
+ Definitive Guide" book. (closes issue ASTERISK-22197) Reported
+ by: Dalius M. Tested by: Dalius M. Patches:
+ asterisk-22197-q-log-exitwithkey.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2901/
+
+2013-10-03 22:51 +0000 [r400469] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Don't ignore expires value in
+ contact header if it lacks semicolon (closes issue
+ ASTERISK-22574) Reported by: Filip Jenicek Patches:
+ chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
+
+2013-10-03 18:25 +0000 [r400393] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_multicast.c: Ensure res_rtp_mutlicast sets SSRC
+ properly This fixes a bug where the SSRC field on multicast RTP
+ can be stuck at 0 which can cause problems for endpoints trying
+ to make sense of incoming streams. (closes issue ASTERISK-22567)
+ Reported by: Simone Camporeale Patches:
+ 22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale
+ (License 6536)
+
+2013-10-02 21:30 +0000 [r400314] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_iax2.c: Cast Integer Argument To Unsigned Char The
+ member reg in the peercnt structure is an unsigned char and
+ peercnt_modify() is expecting an unsigned char argument which
+ gets assigned to peercnt->reg. This patch fixes that by casting
+ the integer argument being passed to peercnt_modify to unsigned
+ char.
+
+2013-09-30 15:19 +0000 [r400137] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Allow
+ Asterisk to retry after 403 on register This adds a global option
+ in chan_sip to allow it to continue attempting registration if a
+ 403 is received, clearing the cached nonce and treating it as a
+ non-fatal response. Normally, this would cause registration
+ attempts to that endpoint to stop. (closes issue ASTERISK-17138)
+ Review: https://reviewboard.asterisk.org/r/2874/ Reported by:
+ Rudi
+
+2013-09-28 22:20 +0000 [r400073-400089] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Correct erroneous lost
+ packet information in RTCP reports RTCP's calculation of the
+ number of lost packets in an RTP stream is based on that stream's
+ sequence number count, the number of received packets, and how
+ many packets we expect to receive. When the SSRC for an RTP
+ stream changes, there can - and almost always will be - a large
+ jump in the next packet's timestamp and sequence number. If we
+ don't reset the number of received packets, sequence number
+ count, and other metrics used by RTCP, the next RR/SR report will
+ use the previous SSRC's values to calculate the lost packet count
+ for the new SSRC - resulting in a very large number of lost
+ packets. This patch modifies res_rtp_asterisk such that, if it
+ detects a SSRC change, it will reset the various values used by
+ the RTCP calculations. From the perspective of RTCP, this appears
+ as a new media stream - which is what it is. Review:
+ https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174)
+ Reported by: Thomas Arimont
+
+ * configure.ac, configure: Add check for openSUSE when detecting
+ bfd library In ASTERISK-17842, some additional library checks
+ were added to the configure script so that the bfd library could
+ be found on CentOS and Fedora systems. As it turns out, openSUSE
+ requires an additional library. This patch adds another check to
+ the configure script for openSUSE that will add that library.
+ Review: https://reviewboard.asterisk.org/r/2885/ (closes issue
+ AST-1169) Reported by: Guenther Kelleter
+
+2013-09-27 21:31 +0000 [r400013] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c, channels/sip/reqresp_parser.c: chan_sip:
+ Increase some scratch buffer sizes dealing with caller id. *
+ Eliminated an unnecessary initialization in check_user_full().
+ (closes issue ASTERISK-22477) Reported by: Michael Shepelev
+
+2013-09-27 17:13 +0000 [r399939] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
+ calls on 200 OKs if no SDP has been received When Asterisk
+ receives a 200 OK in response to an invite, that peer should have
+ sent an SDP at some point by then. If the channel has never
+ received an SDP, media won't have been set and the remote address
+ won't be known. Endpoints in general should not be doing this.
+ This patch makes it so that Asterisk will simply hang up a call
+ if it sends a 200 OK at this point. So far this odd behavior for
+ endpoints has only been observed in tests which involved manually
+ created SIP transactions in SIPp. (closes issue ASTERISK-22424)
+ Reported by: Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/
+
+2013-09-25 20:23 +0000 [r399818] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c, channels/sig_ss7.c: chan_dahdi: CLI "core
+ stop gracefully" has needless delay for PRI and SS7. The PRI and
+ SS7 link control threads are not stopped correctly when the
+ chan_dahdi.so module is unloaded. The link control threads
+ pri_dchannel() and ss7_linkset() are not awakened from a poll()
+ to cancel the thread. * Added a SIGURG signal after requesting
+ the thread cancel to break the link control thread poll()
+ immediately. For SS7 it was slightly worse, the link poll()
+ timeout would always be whatever was the last libss7 scheduled
+ event time used. If no libss7 scheduled event was pending, the
+ thread could run more often than necessary. * Set nextms to 60
+ seconds for the ss7_linkset() poll() if there is no other libss7
+ scheduled event.
+
+2013-09-25 19:25 +0000 [r399794] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Fix Realtime Peer Update Problem When
+ Un-registering And Expires Header In 200ok 1st Issue When a
+ realtime peer sends an un-REGISTER request, Asterisk un-registers
+ the peer but the database table record still has regseconds and
+ fullcontact for the peer. This results in calls attempting to be
+ routed to the peer which is no longer registered. The expected
+ behavior is to get busy/congested when attempting to call an
+ un-registered peer through the dialplan. What was discovered is
+ that we are clearing out the peer's registration in the database
+ in parse_register_contact() when calling expire_register() but
+ then upon returning from parse_register_contact(), update_peer()
+ is run which stores back in the database table regseconds and
+ fullcontact. 2nd Issue The reporter pointed out that the 200 ok
+ being returned by Asterisk after un-registering a peer contains a
+ Contact header with ;expires= and the Expires header is not set
+ to 0. This is actually a regression. Tests were created for this
+ second issue (ASTERISK-22548). The tests have been reviewed and a
+ Ship It! was received on those tests. This patch does the
+ following: * Do not ignore the Expires header value even when it
+ is set to 0. The patch sets the pvt->expiry earlier on in the
+ function so that it is set properly and used. * If pvt->expiry is
+ 0, do not call update_peer since that means the peer has already
+ been un-registered and there is no need to update the database
+ record again since nothing has changed. (closes issue
+ ASTERISK-22428) Reported by: Ben Smithurst Tested by: Ben
+ Smithurst, Michael L. Young Patches:
+ asterisk-22428-rt-peer-update-and-expires-header.diff by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2869/
+
+2013-09-24 20:03 +0000 [r399697] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Prevent some needless breaking
+ of the native IAX2 bridge. * Clean up some twisted code in the
+ iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and
+ AST_CONTROL_SRCCHANGE to a list of frames to prevent the native
+ bridge loop from breaking. * Passing the
+ AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a
+ native IAX2 bridge. (issue ABE-2912) Review:
+ https://reviewboard.asterisk.org/r/2870/
+
+2013-09-19 16:34 +0000 [r399456] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Make direct media reinvites for
+ T38 put Asterisk in the media path Prior to this patch, Asterisk
+ would incorrectly use the previous endpoint addresses in SDP in
+ spite of providing its own port. T38 is never meant to be done
+ through directmedia and Asterisk should always be in the media
+ path for these streams. (closes issue ASTERISK-17273) Reported
+ by: Kevin Stewart (closes issue ASTERISK-18706) Reported by:
+ Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/
+
+2013-10-21 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.24.0 Released.
+
+2013-10-18 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.24.0-rc2 Released.
+
+ * Properly copy/remove the device state cache flag over a masquerade.
+
+ In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that
+ tells the devstate system to not cache states for non-real devices.
+ However, when optimizing away channels (ast_do_masquerade), that flag
+ wasn't copied.
+
+ In my case, using Local devices as queue members created a situation
+ where the endpoint was considered in use, but the state change of the
+ device being available again was ignored (not cached). The endpoint
+ channel was optimized into the (previously) Local channel, but kept
+ the do-not-cache flag. The end result being that the queue member
+ apparently stayed in use forever.
+
+2013-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.24.0-rc1 Released.
+
+2013-09-18 19:54 +0000 [r399402] Kinsey Moore <kmoore at digium.com>
+
+ * main/abstract_jb.c: Fix jitter buffer log file creation This
+ adjusts '/'-to-'#' replacement to replace all instances of '/'
+ instead of just the first to ensure that the jitter buffer log
+ file gets the correct name as per Richard Kenner's suggestion.
+ (closes issue ASTERISK-21036) Reported by: Richard Kenner
+
+2013-09-18 17:15 +0000 [r399351] Matthew Jordan <mjordan at digium.com>
+
+ * build_tools/prep_tarball: Update prep_tarball with new
+ documentation files on the Asterisk wiki This will now pull both
+ a command reference for the version being prepared, as well as an
+ Admin Guide that applies to all versions of Asterisk. (issue
+ ASTERISK-22439) Reported by: Olle Johansson
+
+2013-09-18 01:32 +0000 [r399304] Michael L. Young <elgueromexicano at gmail.com>
+
+ * main/features.c: Fix Segfault When Syntax Of A Line Under
+ [applicationmap] Is Invalid When processing the lines under the
+ [applicationmap] context in features.conf, a segfault occurs from
+ attempting to process a line with an invalid syntax (basically
+ missing most of the arguments). Example: [applicationmap]
+ automon=*6 * This patch moves the checking for empty arguments to
+ before they are accessed. * Also, checked the "todo" comment and
+ removed it. Some applications do not require arguments. (closes
+ issue ASTERISK-22416) Reported by: CGI.NET Tested by: CGI.NET
+ Patches: asterisk-22416-check-syntax-first_v2.diff by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2803
+
+2013-09-16 16:37 +0000 [r399158] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Fix saving the wrong expiry time
+ in astdb. When a new IAX2 client registers, the astdb database is
+ updated with the value of minregexpire defined in iax.conf
+ instead of using the expiry time that is provided by the client.
+ The provided expiry time of the client is updated after inserting
+ the astdb entry. As a consequence, restarting or reloading
+ asterisk creates clients whose registration may expire before
+ they reregister. The clients are therefore unavailable after
+ minregexpire seconds until they reregister. * Move updating of
+ the expiry time to before inserting into the astdb. (closes issue
+ ASTERISK-22504) Reported by: Stefan Wachtler Patches:
+ chan_iax2.c.patch (license #6533) patch uploaded by Stefan
+ Wachtler
+
+2013-09-13 20:47 +0000 [r399098] David M. Lee <dlee at digium.com>
+
+ * main/astobj2.c: Don't write to /tmp/refs when REF_DEBUG is not
+ defined. If MALLOC_DEBUG is enabled, then the debug destructor
+ for the container is used, which would erroneously write to
+ /tmp/refs. This patch only uses the debug destructor if ref_debug
+ is used. (closes issue ASTERISK-22536)
+
+2013-09-13 13:31 +0000 [r399033] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_meetme.c: Fix several crashes in MeetMeAdmin This change
+ ensures that MeetMeAdmin commands requiring a user actually get a
+ user and fixes another issue where an extra dereference could
+ occur for a last-entered user being ejected if a user identifier
+ was also provided. (closes issue ASTERISK-21907) Reported by:
+ Alex Epshteyn Review: https://reviewboard.asterisk.org/r/2844/
+
+2013-09-12 20:09 +0000 [r398937-398977] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Revert
+ r398835 due to failing tests involving originate (issue
+ ASTERISK-22424) Reported by: Jonathan Rose
+
+ * res/res_musiconhold.c: res_musiconhold: Fix reference leaks
+ caused when reloading with REF_DEBUG set Due to a faulty function
+ for debugging reference decrementing, it was possible to reduce
+ the refcount on the wrong object if two moh classes of the same
+ name were in the moh class container. (closes issue
+ ASTERISK-22252) Reported by: Walter Doekes Patches:
+ 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license
+ 6182)
+
+2013-09-12 00:00 +0000 [r398880-398884] Rusty Newton <rnewton at digium.com>
+
+ * apps/app_queue.c: 'queue add member' help text correction You are
+ adding dial strings to the queue, not channels. An aribitrary
+ string could be used, but you are typically referencing a
+ channel. Correcting the command help text. (issue ASTERISK-22263)
+ (closes issue ASTERISK-22263) Reported By: Rusty Newton
+
+ * configs/chan_dahdi.conf.sample: Documentation fix -
+ waitfordialtone is not boolean, it's time in milliseconds
+ Changing text in chan_dahdi.conf sample to be accurate. (issue
+ ASTERISK-22308) (closes issue ASTERISK-22308) Reported By:
+ Malcolm Davenport
+
+2013-09-11 19:39 +0000 [r398835] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: chan_sip: Reject
+ calls without prior SDP on 200 OK If we receive a 200 OK without
+ SDP, we will now check to see if the remote address has been
+ established for that channel's RTP session and if the to tag for
+ that channel has changed from the most recent to tag in a
+ response less than 200. If either a change has been made since
+ the last to-tag was received or the remote address is unset, then
+ we will drop the call. (closes issue ASTERISK-22424) Reported by:
+ Jonathan Rose Review:
+ https://reviewboard.asterisk.org/r/2827/diff/#index_header
+
+2013-09-10 17:53 +0000 [r398757] Richard Mudgett <rmudgett at digium.com>
+
+ * main/xmldoc.c, main/cli.c, funcs/func_dialgroup.c, main/heap.c,
+ main/event.c, res/res_musiconhold.c, main/indications.c,
+ main/asterisk.c: Fix incorrect usages of ast_realloc(). There are
+ several locations in the code base where this is done: buf =
+ ast_realloc(buf, new_size); This is going to leak the original
+ buf contents if the realloc fails. Review:
+ https://reviewboard.asterisk.org/r/2832/
+
+2013-09-10 17:47 +0000 [r398748-398752] David M. Lee <dlee at digium.com>
+
+ * utils/check_expr.c: Fixed utils directory breakage from r398748,
+ this time with extra hate.
+
+ * utils/check_expr.c, utils/ael_main.c, utils/conf2ael.c: Fixed
+ utils directory breakage from r398648
+
+2013-09-09 23:15 +0000 [r398703] Richard Mudgett <rmudgett at digium.com>
+
+ * main/astmm.c: MALLOC_DEBUG: Change fence magic number to be
+ completely different from the freed magic number. Race conditions
+ between freeing a nul terminated string and ast_strdup()'ing it
+ are more likely to be detected if the fence and freed magic
+ numbers are completely different.
+
+2013-09-09 19:56 +0000 [r398648] David M. Lee <dlee at digium.com>
+
+ * main/utils.c, include/asterisk/lock.h, main/lock.c: Fix
+ DEBUG_THREADS when lock is acquired in __constructor__ This patch
+ fixes some long-standing bugs in debug threads that were
+ exacerbated with recent Optional API work in Asterisk 12. With
+ debug threads enabled, on some systems, there's a lock ordering
+ problem between our mutex and glibc's mutex protecting its module
+ list (Ubuntu Lucid, glibc 2.11.1 in this instance). In one
+ thread, the module list will be locked before acquiring our
+ mutex. In another thread, our mutex will be locked before locking
+ the module list (which happens in the depths of calling
+ backtrace()). This patch fixes this issue by moving backtrace()
+ calls outside of critical sections that have the mutex acquired.
+ The bigger change was to reentrancy tracking for
+ ast_cond_{timed,}wait, which wrongly assumed that waiting on the
+ mutex was equivalent to a single unlock (it actually suspends all
+ recursive locks on the mutex). (closes issue ASTERISK-22455)
+ Review: https://reviewboard.asterisk.org/r/2824/
+
+2013-09-06 20:56 +0000 [r398523-398576] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_jabber.c: Commit the remainder of r398523 This is a
+ missing part of the commit in revision 398523 that corrects the
+ name of a variable. (issue ASTERISK-22435)
+
+ * res/res_jabber.c: Fix Jabber/XMPP distributed MWI The mailbox and
+ context are swapped on the receiving end for all users of Jabber
+ and XMPP distributed MWI in Asterisk 1.8 and all more recent
+ versions. This swaps those values to be correct when publishing
+ to the internal event system from Jabber/XMPP distributed MWI
+ state. (closes issue ASTERISK-22435) Reported by: abelbeck Tested
+ by: Michael Keuter Patches:
+ asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by
+ abelbeck asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch
+ uploaded by abelbeck
+
+2013-09-05 19:00 +0000 [r398235-398456] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Reduce indentation in
+ __attempt_transmit(). * Reduce indentation in
+ __attempt_transmit(). * Don't update the static last error time
+ variable every time in __schedule_action() and socket_read().
+
+ * channels/chan_iax2.c: chan_iax2: Fix stray reference to worker
+ thread idle_list. * Fix stray reference to idle_list in
+ cleanup_thread_list(). This may be the reason for the note in
+ iax2_process_thread() about threads not being removed from the
+ task lists. * Move cleanup_thread_list(&idle_list) to after the
+ other lists are cleaned up.
+
+ * channels/chan_iax2.c: chan_iax2: Fix bridgecallno deadlock
+ avoidance. * Fix bridgecallno deadlock avoidance. When doing
+ deadlock avoidance, you need to retest the status of values for
+ each loop to see if you still need the lock for bridgecallno. *
+ As a safety check, after acquiring the bridgecallno lock you
+ should check if iaxs[bridgecallno] is NULL just like the current
+ callno checks. * Move setting thread->iostate to IAX_IOSTATE_IDLE
+ to after processing any deferred frames to ensure that the
+ iostate is IDLE when it is placed back into the idle list.
+ defer_full_frame() tries to ensure iax2_process_thread() wakes up
+ to process the frame.
+
+ * channels/iax2-parser.c: chan_iax2: Add missing control frame
+ names to debug frame decode output. (Part 2)
+
+ * channels/iax2-parser.c: chan_iax2: Add missing control frame
+ names to debug frame decode output.
+
+ * channels/chan_misdn.c: chan_misdn: Fix misdn debug output printed
+ with arbitrary verbose levels. Fix the misdn debug output to
+ remote consoles. chan_misdn uses ast_console_puts() which doesn't
+ know about verbose levels. Better to use ast_verbose() instead.
+ Without this patch the misdn debug messages are appended to the
+ verbose level which ever was set by the message sent to the
+ console before, i.e. any undefined level. (closes issue AST-1218)
+ Reported by: Guenther Kelleter Patches: misdnlog.patch (license
+ #6372) patch uploaded by Guenther Kelleter
+
+2013-09-02 07:24 +0000 [r398167] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * cel/cel_custom.c: Be a little more verbose when loading
+ cel_custom.conf. Review: https://reviewboard.asterisk.org/r/2805/
+
+2013-08-30 18:55 +0000 [r398021-398102] Kevin Harwell <kharwell at digium.com>
+
+ * channels/chan_sip.c, main/config.c, res/res_security_log.c: Fix
+ various memory leaks main/config.c - cleanup cache fie includes
+ res/res_security_log.c - unregister logger level
+ channesl/chan_sip.c - cleanup io context and notify_types (closes
+ issues ASTERISK-22378) Reported by: Corey Farrell Patches:
+ config_shutdown.patch uploaded by coreyfarrell (license 5909)
+ res_security_log.patch uploaded by coreyfarrell (license 5909)
+ chan_sip-1.8.patch uploaded by coreyfarrell (license 5909)
+
+ * main/manager.c, res/res_agi.c: Memory leak fix
+ ast_xmldoc_printable returns an allocated block that must be
+ freed by the caller. Fixed manager.c and res_agi.c to stop
+ leaking these results. (closes issue ASTERISK-22395) Reported by:
+ Corey Farrell Patches: manager-leaks-1.8.patch uploaded by
+ coreyfarrell (license 5909) res_agi-xmldoc-leaks.patch uploaded
+ by coreyfarrell (license 5909)
+
+ * main/features.c: Fix memory leak Fixed a features.c test that
+ leaked a reference to a parked call. This caused chancount to
+ never reach 0, so graceful shutdown stops. Also added an
+ unregister test. (closes issue ASTERISK-22413) Reported by: Corey
+ Farrell Patches: features-TEST_FRAMEWORK.patch uploaded by
+ coreyfarrell (license 5909)
+
+2013-08-30 16:46 +0000 [r398018] Richard Mudgett <rmudgett at digium.com>
+
+ * tests/test_substitution.c: test_substituition: Fix failed test
+ reporting to actually report failure. You cannot put the "Testing
+ <blah> pass/fail" on a single line before actually performing the
+ test. Now any additional failure information is logged before the
+ test pass/fail announcement. * Added an additional CDR(answer,u)
+ test.
+
+2013-08-27 17:55 +0000 [r397710-397756] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: AST-2013-005: Fix crash caused by invalid
+ SDP If the SIP channel driver processes an invalid SDP that
+ defines media descriptions before connection information, it may
+ attempt to reference the socket address information even though
+ that information has not yet been set. This will cause a crash.
+ This patch adds checks when handling the various media
+ descriptions that ensures the media descriptions are handled only
+ if we have connection information suitable for that media. Thanks
+ to Walter Doekes, OSSO B.V., for reporting, testing, and
+ providing the solution to this problem. (closes issue
+ ASTERISK-22007) Reported by: wdoekes Tested by: wdoekes patches:
+ issueA22007_sdp_without_c_death.patch uploaded by wdoekes
+ (License 5674)
+
+ * channels/chan_sip.c: AST-2013-004: Fix crash when handling ACK on
+ dialog that has no channel A remote exploitable crash
+ vulnerability exists in the SIP channel driver if an ACK with SDP
+ is received after the channel has been terminated. The handling
+ code incorrectly assumed that the channel would always be
+ present. This patch adds a check such that the SDP will only be
+ parsed and applied if Asterisk has a channel present that is
+ associated with the dialog. Note that the patch being applied was
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