[asterisk-commits] bebuild: tag 11.7.0-rc1 r402106 - /tags/11.7.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Oct 28 09:00:39 CDT 2013
Author: bebuild
Date: Mon Oct 28 09:00:37 2013
New Revision: 402106
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=402106
Log:
Importing files for 11.7.0-rc1 release.
Added:
tags/11.7.0-rc1/.lastclean (with props)
tags/11.7.0-rc1/.version (with props)
tags/11.7.0-rc1/ChangeLog (with props)
Added: tags/11.7.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/11.7.0-rc1/.lastclean?view=auto&rev=402106
==============================================================================
--- tags/11.7.0-rc1/.lastclean (added)
+++ tags/11.7.0-rc1/.lastclean Mon Oct 28 09:00:37 2013
@@ -1,0 +1,1 @@
+40
Propchange: tags/11.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/11.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/11.7.0-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/11.7.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/11.7.0-rc1/.version?view=auto&rev=402106
==============================================================================
--- tags/11.7.0-rc1/.version (added)
+++ tags/11.7.0-rc1/.version Mon Oct 28 09:00:37 2013
@@ -1,0 +1,1 @@
+11.7.0-rc1
Propchange: tags/11.7.0-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/11.7.0-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/11.7.0-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/11.7.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/11.7.0-rc1/ChangeLog?view=auto&rev=402106
==============================================================================
--- tags/11.7.0-rc1/ChangeLog (added)
+++ tags/11.7.0-rc1/ChangeLog Mon Oct 28 09:00:37 2013
@@ -1,0 +1,27613 @@
+2013-10-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.7.0-rc1 Released.
+
+2013-10-25 23:32 +0000 [r401960-402042] 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. (closes issue
+ ASTERISK-21464) Reported by: Kevin Stewart Review:
+ https://reviewboard.asterisk.org/r/2894/ ........ Merged
+ revisions 402000 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8 Test shows
+ rtpmap:119 being copied per this change, but is not in sip invite
+
+ * 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/ ........ Merged
+ revisions 401959 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-25 17:29 +0000 [r401896-401935] 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) ........ Merged revisions 401914
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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 ........ Merged
+ revisions 401895 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-25 16:05 +0000 [r401833-401884] Kevin Harwell <kharwell at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Allow a sip peer to accept both
+ AVP and AVPF calls Adapts the behaviour of avpf to only impact
+ the format of outgoing calls. For inbound calls, both AVP and
+ AVPF calls will be accepted regardless of the value of avpf in
+ the configuration. (closes issue ASTERISK-22005) Reported by:
+ Torrey Searle Patches: optional_avpf_trunk.patch uploaded by
+ tsearle (license 5334)
+
+ * main/logger.c: Logging: Logging types ignored after specifying a
+ verbose level If one specified a verbose level within a logging
+ facility in logger.conf then any component after it was ignored.
+ Fixed so all values are correctly read. (closes issue
+ ASTERISK-22456) Reported by: Kevin Harwell
+
+2013-10-24 20:33 +0000 [r401620-401830] Jonathan Rose <jrose at digium.com>
+
+ * /, 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)
+ ........ Merged revisions 401829 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged revisions 401790 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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 ........ Merged revisions
+ 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged revisions 401781 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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)
+ ........ Merged revisions 401743 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * utils/clicompat.c, channels/chan_dahdi.c, codecs/ilbc/doCPLC.c,
+ main/data.c, /, main/app.c, main/asterisk.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) ........ Merged revisions 401704 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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) ........ Merged revisions 401660 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_rtp_asterisk.c, /, main/translate.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/ ........ Merged
+ revisions 401619 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-23 16:46 +0000 [r401579] 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 ........ Merged revisions 401577 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-23 15:22 +0000 [r401538] 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)
+ ........ Merged revisions 401537 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-23 11:11 +0000 [r401498] 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) ........ Merged revisions 401497 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-22 22:42 +0000 [r401446] 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 ........ Merged revisions 401445 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-22 19:02 +0000 [r401379-401433] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_queue.c: app_queue: Fix CLI "queue remove member"
+ queue_log entry. The queue_log entry resulting from CLI "queue
+ remove member" when log_membername_as_agent is enabled is wrong.
+ It always uses the interface name instead of the member name in
+ the queue_log entry. * Get the queue member before removing it
+ from the queue so the member name is available for the queue_log
+ entry. (closes issue ASTERISK-21826) Reported by: Oscar Esteve
+ Patches: fix_membername.diff (license #6505) patch uploaded by
+ Oscar Esteve (modified to fix potential ref leak)
+
+ * 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. ........ Merged revisions 401378
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-21 19:46 +0000 [r401326] 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) ........ Merged revisions 401325 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-18 15:11 +0000 [r401182] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Remove Port Restriction When Checking For
+ NAT When trying to determine if a peer is behind NAT, we should
+ not be using the ports when comparing addresses. This patch
+ removes the port from being checked and just useds the addresses
+ now. (closes issue ASTERISK-22729) Reported by: Michael L. Young
+ Tested by: Michael L. Young Patches:
+ asterisk-remove-using-port-for-nat-check.diff uploaded by Michael
+ L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2927/
+
+2013-10-18 14:43 +0000 [r401179] 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/ ........ Merged
+ revisions 401178 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-17 20:32 +0000 [r401167] Michael L. Young <elgueromexicano at gmail.com>
+
+ * channels/chan_sip.c: Fix Setting A chan_sip Dialog's
+ SIP_NAT_FORCE_RPORT Flag A condition was added in a commit to fix
+ ASTERISK-21374, that, if the SIP_PAGE3_NAT_AUTO_RPORT flag was
+ set, to then copy a peer's SIP_NAT_FORCE_RPORT flag to the
+ dialog. This condition should not have been there since it
+ assumed that if Asterisk is in an environment where NAT is
+ involved, that the auto_* nat settings or force_rport setting
+ would be on in the global settings. If the nat setting in the
+ global setting is set to 'nat=no' and then turned on for peers
+ (which is not quite the recommended way, although it is allowed)
+ this flag is never copied to the dialog resulting in problems
+ like, REGISTER replies going to the wrong port. This patch
+ removes this conditional check and will now always use the peer's
+ flag which by this point in the code the checks on whether the
+ peer is behind NAT or not (if using auto_force_rport) have
+ already been run. (closes issue ASTERISK-22236) Reported by:
+ Filip Frank Tested by: Michael L. Young Patches:
+ asterisk-2236-always-set-rport.diff uploaded by Michael L. Young
+ (license 5026) Review: https://reviewboard.asterisk.org/r/2919/
+
+2013-10-17 15:36 +0000 [r401120] Kinsey Moore <kmoore at digium.com>
+
+ * /, res/res_xmpp.c, 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)
+ asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded
+ by abelbeck (License 5903) ........ Merged revisions 401119 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-16 11:52 +0000 [r401076] 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/ ........ Merged
+ revisions 401049 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-15 19:57 +0000 [r401016] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_iax2.c: chan_iax2: Fix channel left locked in off
+ nominal code path.
+
+2013-10-15 14:58 +0000 [r400971] 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 ........ Merged revisions 400970 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-14 21:44 +0000 [r400909] 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 ........ Merged revisions 400907 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-14 21:42 +0000 [r400908] 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) ........ Merged revisions 400906
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-08 22:27 +0000 [r400768] 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) ........ Merged revisions 400767 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-08 20:14 +0000 [r400723-400741] Richard Mudgett <rmudgett at digium.com>
+
+ * UPGRADE.txt, apps/app_confbridge.c,
+ apps/confbridge/conf_config_parser.c,
+ configs/confbridge.conf.sample,
+ apps/confbridge/include/confbridge.h: app_confbridge: Can now set
+ the language used for announcements to the conference. ConfBridge
+ now has the ability to set the language of announcements to the
+ conference. The language can be set on a bridge profile in
+ confbridge.conf or by the dialplan function
+ CONFBRIDGE(bridge,language)=en. (closes issue ASTERISK-19983)
+ Reported by: Jonathan White Patches: M19983_rev2.diff (license
+ #5138) patch uploaded by junky (modified) Tested by: rmudgett
+
+ * apps/confbridge/conf_config_parser.c: app_confbridge: Fix
+ duplicate default_user profile. * Fixed looking in the wrong
+ profiles container to see if the default_user profile is already
+ created in verify_default_profiles(). The bridge profile
+ container is never going to hold user profiles. :)
+
+2013-10-08 18:18 +0000 [r400681-400697] Kinsey Moore <kmoore at digium.com>
+
+ * 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 ........ Merged revisions 400694 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_rtp_asterisk.c: Fix STUN crash when using IPv6 any
+ address Ensure that when chan_sip binds to the IPv6 any address
+ ([::]), IPv4 candidates are also added. (closes issue
+ ASTERISK-21917) Reported by: Torrey Searle Patches:
+ 0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License
+ 5334)
+
+2013-10-06 17:09 +0000 [r400623] 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/ ........ Merged
+ revisions 400622 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-03 22:59 +0000 [r400470] 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)
+ ........ Merged revisions 400469 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-03 19:22 +0000 [r400394-400421] Kinsey Moore <kmoore at digium.com>
+
+ * main/security_events.c: Fix security events for AMI invalid
+ password In r337595, additional security events were added for
+ chan_sip authentication failures. The new IEs added to the
+ existing invalid password event were defined as required IEs, but
+ existing users of the event did not set the new IEs and could not
+ since they didn't apply to existing uses. They are now marked as
+ optional IEs. (closes issue ASTERISK-22578) Reported by: Matt
+ Jordan
+
+ * 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) ........ Merged revisions 400393 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-02 21:31 +0000 [r400315] 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. ........ Merged revisions 400314 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-02 17:36 +0000 [r400279] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * Makefile, doc/astdb2sqlite3.8 (added), doc/astdb2bdb.8 (added):
+ man pages for astdb2bdb and astdb2sqlite3 Review:
+ https://reviewboard.asterisk.org/r/2898/
+
+2013-09-30 15:26 +0000 [r400140] Kinsey Moore <kmoore at digium.com>
+
+ * UPGRADE.txt, configs/sip.conf.sample, /, channels/chan_sip.c:
+ 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 ........ Merged revisions 400137 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-28 22:21 +0000 [r400075-400093] 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 ........ Merged revisions 400089 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, configure, configure.ac: 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 ........ Merged
+ revisions 400073 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-27 21:35 +0000 [r400014] 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
+ ........ Merged revisions 400013 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-27 17:24 +0000 [r399962] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/include/sip.h: 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/ ........ Merged
+ revisions 399939 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-25 20:28 +0000 [r399834] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_ss7.c, channels/chan_dahdi.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. ........ Merged revisions 399818 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-25 19:27 +0000 [r399795] 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/ ........ Merged
+ revisions 399794 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-24 20:20 +0000 [r399708] 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/ ........ Merged
+ revisions 399697 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-20 22:35 +0000 [r399564] Kinsey Moore <kmoore at digium.com>
+
+ * main/config_options.c: Ensure global types in the config
+ framework are initialized If a config object was allocated but
+ one of its global objects was never encountered, then the global
+ object's defaults were never applied. Ensure that global objects
+ are initialized properly upon allocation instead of on
+ configuration. Review: https://reviewboard.asterisk.org/r/2866/
+
+2013-09-20 14:23 +0000 [r399513] Kevin Harwell <kharwell at digium.com>
+
+ * main/logger.c: Fix memory leak in logger. Fixed a memory leak
+ discovered in the logger where a temporary string buffer was not
+ being freed. (closes issue ASTERISK-22540) Reported by: John
+ Hardin
+
+2013-09-19 16:45 +0000 [r399457] 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/
+ ........ Merged revisions 399456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-10-21 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.6.0 Released.
+
+2013-10-18 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.6.0-rc2 Released.
+
+ * Remove Port Restriction When Checking For NAT
+
+ When trying to determine if a peer is behind NAT, we should not be
+ using the ports when comparing addresses.
+
+ This patch removes the port from being checked and just useds the
+ addresses now.
+
+ * 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.
+
+ * Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag
+
+ A condition was added in a commit to fix ASTERISK-21374, that, if the
+ SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's
+ SIP_NAT_FORCE_RPORT flag to the dialog. This condition should not
+ have been there since it assumed that if Asterisk is in an
+ environment where NAT is involved, that the auto_* nat settings or
+ force_rport setting would be on in the global settings. If the nat
+ setting in the global setting is set to 'nat=no' and then turned on
+ for peers (which is not quite the recommended way, although it is
+ allowed) this flag is never copied to the dialog resulting in
+ problems like, REGISTER replies going to the wrong port.
+
+ This patch removes this conditional check and will now always use the
+ peer's flag which by this point in the code the checks on whether the
+ peer is behind NAT or not (if using auto_force_rport) have already
+ been run.
+
+ * Fix memory leak in logger
+
+ Fixed a memory leak discovered in the logger where a temporary string
+ buffer was not being freed.
+
+2013-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.6.0-rc1 Released.
+
+2013-09-18 23:36 +0000 [r399442] Richard Mudgett <rmudgett at digium.com>
+
+ * main/udptl.c: UDPTL: Backport some fixes from v12 that should be
+ in v11. Backported the following as applied to udptl.c: *
+ -r398020 Fixup udpdl defaults if config file not present. *
+ -r398533 Fixup improper use of ao2_global_obj_replace().
+
+2013-09-18 19:55 +0000 [r399403] 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
+ ........ Merged revisions 399402 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2013-09-18 17:22 +0000 [r399353-399373] 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 ........ Merged
+ revisions 399351 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * bridges/bridge_softmix.c: Add a WARNING in bridge_softmix when a
+ timing module isn't loaded If bridge_softmix fails to be created
+ because no timing source is present in Asterisk, this will
+ currently fail gracefully but with (most likely) a generic error
+ message by whatever module tried to create the softmix bridge.
+ This patch adds a more explicit warning so you can actually
+ diagnose and fix the problem. Review:
+ https://reviewboard.asterisk.org/r/2857/
+
+2013-09-18 01:34 +0000 [r399305] 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
[... 26910 lines stripped ...]
More information about the asterisk-commits
mailing list