[asterisk-commits] bebuild: tag 1.8.8.0-rc1 r339568 - /tags/1.8.8.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Oct 5 16:37:52 CDT 2011
Author: bebuild
Date: Wed Oct 5 16:37:48 2011
New Revision: 339568
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=339568
Log:
Importing files for 1.8.8.0-rc1 release.
Added:
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tags/1.8.8.0-rc1/.version (with props)
tags/1.8.8.0-rc1/ChangeLog (with props)
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--- tags/1.8.8.0-rc1/ChangeLog (added)
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+2011-10-05 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.8.0-rc1 Released.
+
+2011-10-05 21:30 +0000 [r339566] Leif Madsen <lmadsen at digium.com>
+
+ * build_tools/prep_tarball: Update prep_tarball script to download
+ pre-exported documentation. I've updated the prep_tarball script
+ to now download the pre-exported documentation from the Asterisk
+ wiki. This will give us more control over what is being included
+ in the tarball releases, and will make both the PDF and HTML
+ exported documentation look much better (especially when viewing
+ from a console). (Closes issue ASTERISK-18677)
+
+2011-10-05 17:01 +0000 [r339506-339511] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c: Fix Dial F option notes formatting.
+
+ * main/manager.c: Fix XML error in AMI action Challenge.
+
+2011-10-05 16:31 +0000 [r339505] Matthew Nicholson <mnicholson at digium.com>
+
+ * res/res_fax.c: The app name in the documentation must match what
+ we register the application as.
+
+2011-10-05 16:26 +0000 [r339406-339504] Richard Mudgett <rmudgett at digium.com>
+
+ * main/manager.c: Add missing documentation of required AMI action
+ Challenge AuthType header. (closes issue ASTERISK-18554) Reported
+ by: Vlad Povorozniuc Patches:
+ __20110919-manager-challenge-docs.patch.txt (license #4999) patch
+ uploaded by Leif Madsen
+
+ * Makefile: Make always create the MOH directory
+ (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
+ by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
+ #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
+ Keuter
+
+2011-10-04 19:33 +0000 [r339297-339352] Jonathan Rose <jrose at digium.com>
+
+ * main/say.c: Removes improper use of sound 'and' in German
+ language mode from application saynumber Asterisk would say 'Five
+ hundert und sechs und zwanzig' instead of 'Five hundert sechs und
+ zwanzig'... which is both weird sounding and wrong. This patch
+ makes sure Asterisk will only say the 'and' word between the
+ single digit and double digit places. (closes issue
+ ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
+ upstream_germand_no_and.diff (License #5402) uploaded by Lionel
+ Elie Mamane
+
+ * res/res_jabber.c: Reverting revision 333265 due to component
+ connection problems it introduces. I'm going to attempt some
+ generic res_jabber cleanup and come up with a new fix for this
+ problem, but first it seems prudent to remove this rather broad
+ attempt to fix it and instead approach this problem either from
+ the same angle but looking only at canceling (or possibly
+ rescheduling) the send when we absolutely know it will cause a
+ segfault or, if that can't be easily accomplished, strictly from
+ the devstate side of things. Also, I'm pretty sure a lot of the
+ code in res_jabber isn't thread safe. (issue ASTERISK-18626)
+ (issue ASTERISK-18078)
+
+2011-10-04 11:44 +0000 [r339244] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c: fix forget declaration in previous
+ change
+
+2011-10-03 20:12 +0000 [r339144-339147] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_sip.c: Remove duplicated Maxforwards line in AMI
+ output. (Closes issue ASTERISK-18637) Reported by: Jacek
+ Konieczny Patches: asterisk-sipshowpeer.patch (License #6298)
+ uploaded by Jacek Konieczny
+
+ * apps/app_dial.c: Make documentation for Dial() options 'F' and
+ 'F()' more clear. (Closes issue ASTERISK-18646) Reported by:
+ Physis Heckman Tested by: Richard Mudgett
+
+2011-10-03 18:42 +0000 [r339087] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/memheap.c: destroy memheap mutex properly
+ before memheap deleted (fix memory leak occured after r304950
+ changes with DEBUG_THREAD compile option)
+
+2011-10-03 18:40 +0000 [r339086] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c, main/file.c: Properly ignore
+ AST_CONTROL_UPDATE_RTP_PEER in more places After the change in
+ r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a
+ re-invite happens. If we receive a re-invite from a device the
+ waitstream_core was not aware of the new control frame and would
+ drop the call. (closes issue ASTERISK-18610) Reported by:
+ Kristijan_Vrban
+
+2011-09-30 22:05 +0000 [r338800] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix segfault in analog_ss_thread() not
+ checking ast_read() for NULL. NOTE: The problem was reported
+ against v1.6.2. It is unlikely to ever happen on v1.8 and above
+ since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
+ version in sig_analog.c has largely replaced it. (closes issue
+ ASTERISK-18648) Reported by: Stephan Bosch Patches:
+ jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Stephan Bosch
+
+2011-09-30 18:54 +0000 [r338718] Jonathan Rose <jrose at digium.com>
+
+ * configs/queues.conf.sample: Adds documentation for
+ QueueMemberStatus event generation
+
+2011-09-30 16:27 +0000 [r338663] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: Fix formatting of AMI header for SIP show
+ peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
+ issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
+ asterisk-sipshowpeer_response_end.patch (license #6298) patch
+ uploaded by Jacek Konieczny
+
+2011-09-30 09:31 +0000 [r338609] TransNexus OSP Development <support at transnexus.com>
+
+ * apps/app_osplookup.c, configure.ac: Remove r338137 and r338138.
+
+2011-09-29 21:12 +0000 [r338555] Paul Belanger <pabelanger at digium.com>
+
+ * tests/test_linkedlists.c, tests/test_amihooks.c,
+ tests/test_security_events.c, tests/test_locale.c,
+ tests/test_logger.c, tests/test_dlinklists.c: Test modules should
+ depend on the TEST_FRAMEWORK flag
+
+2011-09-29 20:54 +0000 [r338551] Jason Parker <jparker at digium.com>
+
+ * tests/test_db.c, tests/test_netsock2.c: Test modules have a
+ support level of core.
+
+2011-09-29 18:31 +0000 [r338492] Leif Madsen <lmadsen at digium.com>
+
+ * channels/chan_sip.c: Update documentation for SIP_HEADER. The
+ SIP_HEADER function only works on the the initial SIP INVITE. The
+ documentation was updated in trunk, but not in 1.8 or 10, so I'm
+ making them match. (Closes issue ASTERISK-18640)
+
+2011-09-29 12:13 +0000 [r338416] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/sip/include/sip.h, channels/chan_sip.c: The rtptimeout
+ setting is ignored on a per peer basis. Not only is the
+ rtptimeout ignored in some cases but rtpkeepalive and
+ rtpholdtimeout is affected. this commit also removes
+ rtptimeout/rtpholdtimeout on text rtp. (closes issue
+ ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
+
+2011-09-28 22:35 +0000 [r338235-338322] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Make duplicate call ptr warning message more
+ helpful. * Adds the value of the call ptr to the duplicate call
+ ptr message to help trace why there is a duplicate call ptr.
+
+ * include/asterisk/logger.h: Fix inconsistency in
+ LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue
+ ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch
+ (license #6278) patch uploaded by Luke H
+
+2011-09-28 20:52 +0000 [r338227] Jason Parker <jparker at digium.com>
+
+ * tests/test_db.c, tests/test_netsock2.c, build_tools/cflags.xml,
+ channels/chan_usbradio.c, build_tools/cflags-devmode.xml,
+ agi/agi.xml, utils/utils.xml, build_tools/embed_modules.xml: Add
+ support levels to non-module sections of menuselect (cflags,
+ utils, etc).
+
+2011-09-28 20:24 +0000 [r338224] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: Fix chan_dahd compiling with gcc 4.6 when
+ PRI and SS7 not present. (closes issue ASTERISK-18357) Reported
+ by: Matthew Nicholson
+
+2011-09-28 07:28 +0000 [r338137-338138] TransNexus OSP Development <support at transnexus.com>
+
+ * configure.ac: Updated for checking OSP Toolkit version 4.0.0.
+
+ * apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.
+
+2011-09-27 20:10 +0000 [r338084] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_macro.c: Upgrade app_macro to core
+
+2011-09-26 19:30 +0000 [r337973] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/channel.h, main/cel.c, main/manager.c,
+ funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
+ main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_manager.c,
+ cdr/cdr_custom.c, apps/app_voicemail.c, apps/app_dial.c,
+ main/pbx.c, cdr/cdr_sqlite3_custom.c, cdr/cdr_syslog.c,
+ tests/test_gosub.c, include/asterisk/cel.h: Fix deadlock when
+ using dummy channels. Dummy channels created by
+ ast_dummy_channel_alloc() should be destoyed by
+ ast_channel_unref(). Using ast_channel_release() needlessly grabs
+ the channel container lock and can cause a deadlock as a result.
+ * Analyzed use of ast_dummy_channel_alloc() and made use
+ ast_channel_unref() when done with the dummy channel. (Primary
+ reason for the reported deadlock.) * Made
+ app_dial.c:dial_exec_full() not call ast_call() holding any
+ channel locks. Chan_local could not perform deadlock avoidance
+ correctly. (Potential deadlock exposed by this issue. Secondary
+ reason for the reported deadlock since the held lock was part of
+ the deadlock chain.) * Fixed some uses of
+ ast_dummy_channel_alloc() not checking the returned channel
+ pointer for failure. * Fixed some potential chan=NULL pointer
+ usage in func_odbc.c. Protected by testing the bogus_chan value.
+ * Fixed needlessly clearing a 1024 char auto array when setting
+ the first char to zero is enough in manager.c:action_getvar().
+ (closes issue ASTERISK-18613) Reported by: Thomas Arimont
+ Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
+ uploaded by rmudgett Tested by: Thomas Arimont
+
+2011-09-23 19:14 +0000 [r337839-337898] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * contrib/init.d/rc.archlinux.asterisk: Spelling fix
+
+ * apps/app_queue.c: Make sure a CDR is on the stack for call in the
+ Queue. Only let update_cdr act on the last CDR in the stack. In
+ some circumstances [Attended transfer to queue] a CDR record is
+ not inserted for this call where it should. (closes issue
+ ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
+
+2011-09-23 00:44 +0000 [r337774] Russell Bryant <russell at digium.com>
+
+ * configs/res_pktccops.conf.sample: Comment out entries in sample
+ res_pktccops.conf. With these options enabled, they can cause
+ Asterisk to freak out by SYN flooding a network and eating the
+ CPU. Obviously it would be good to fix the code so that this
+ can't happen, but we can at least change the default
+ configuration so it doesn't happen. This was reported downstream
+ to the Fedora issue tracker:
+ https://bugzilla.redhat.com/show_bug.cgi?id=658431
+
+2011-09-22 21:29 +0000 [r337720] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Made ISDN not add numbering plan prefix
+ strings to empty numbers. When the Caller-ID is restricted, the
+ expected behavior is for the Caller-ID to be blank. In
+ chan_dahdi, the national prefix is placed onto the Caller-ID
+ number even if it is restricted (empty) causing the Caller-ID to
+ be the national prefix rather than blank. This behavior was lost
+ when sig_pri was extracted from chan_dahdi. * Made not add prefix
+ strings to empty connected line, calling, and ANI number strings.
+ (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches:
+ jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Kris Shaw
+
+2011-09-22 11:39 +0000 [r337430-337541] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * res/res_srtp.c: Add warned to ast_srtp to prevent errors on each
+ frame from libsrtp The first 9 frames are not reported as some
+ devices dont use srtp from first frame these are suppresed. the
+ warning is then output only once every 100 frames.
+
+ * channels/chan_h323.c: If IP address is used in chan_h323 host
+ parameter of peer configuration. module tries to resolve IP
+ address to IP address and fails. Simple fix to set family of
+ socket this is a hangover from ipv6 changes. (closes issue
+ ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500)
+
+ * main/channel.c: Its possible to loose audio on ast_write when the
+ channel is not transcoded correctly. in the case of DAHDI the
+ channel is hungup. This patch tries to "fix" the problem and make
+ the channel compatiable and warn the user of this problem. Please
+ note there is a underlying problem with codec negotion this does
+ not fix the problem it does try to rectify it and prevent loss of
+ service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
+ issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
+ ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
+ (issue ASTERISK-18422)
+
+2011-09-21 21:18 +0000 [r337325-337353] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_voicemail.c: More silly spacing changes
+
+ * apps/app_voicemail.c: Dumb little spacing fix.
+
+ * funcs/func_curl.c: Escape commas in keys and values, when keys
+ and values are enumerated by commas. Review:
+ https://reviewboard.asterisk.org/r/1433
+
+2011-09-20 22:38 +0000 [r337118] Matthew Jordan <mjordan at digium.com>
+
+ * main/app.c, apps/app_followme.c, apps/app_voicemail.c,
+ apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
+ apps/app_minivm.c: Fix for incorrect voicemail duration in
+ external notifications This patch fixes an issue where the
+ voicemail duration was being reported with a duration
+ significantly less than the actual sound file duration.
+ Voicemails that contained mostly silence were reporting the
+ duration of only the sound in the file, as opposed to the
+ duration of the file with the silence. This patch fixes this by
+ having two durations reported in the __ast_play_and_record family
+ of functions - the sound_duration and the actual duration of the
+ file. The sound_duration, which is optional, now reports the
+ duration of the sound in the file, while the actual full duration
+ of the file is reported in the duration parameter. This allows
+ the voicemail applications to use the sound_duration for minimum
+ duration checking, while reporting the full duration to external
+ parties if the voicemail is kept. (issue ASTERISK-2234) (closes
+ issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
+ House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/1443
+
+2011-09-20 22:18 +0000 [r337115] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: Update RedHat Init script to
+ work with Heartbeat. The current RedHat init script was not LSB
+ compatible. This change will make it LSB compatible so that it
+ can work correctly with Heartbeat. (Closes issue ASTERISK-18253)
+ Reported by: c0rnoTa
+
+2011-09-20 21:04 +0000 [r337061] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_pbx.c, main/pbx.c: Make CANMATCH with the new pattern
+ match engine behave more like the old one When checking an
+ extension for E_CANMATCH using the new extension matching
+ algorithm, an exact match was not returned as a possible match
+ resulting in the queue failing to allow a caller to exit on DTMF.
+ This removes the requirement that an extension be longer than
+ acquired digits for an E_CANMATCH operation to succeed. (closes
+ issue ASTERISK-18044) Review:
+ https://reviewboard.asterisk.org/r/1367/
+
+2011-09-20 19:10 +0000 [r336977-337007] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_ss7.c: Check if a channel was created before using
+ the pointer in sig_ss7_new_ast_channel(). Fixes the crash in
+ ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
+ libss7 access lock protection. * Prevent cancelling the
+ ss7_linkset() thread at inoportune times just like the
+ pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
+ Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
+ patch uploaded by rmudgett (attached to related ASTERISK-17966)
+
+ * channels/sig_ss7.c: Fix deadlock from not releasing SS7 linkset
+ lock. sig_ss7_hangup() failed to release the SS7 linkset lock if
+ the call had the alreadyhungup flag set. * Made unlock the SS7
+ linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
+ set. * Made ss7_start_call() not hold any locks while creating
+ the channel for an incoming call to prevent deadlock. * Made
+ ss7_grab() a void function, since it could never fail, to
+ simplify calling code. * Made obtain the channel lock to do
+ softhangup in some places. Patches: jira_ast_668_v1.8.patch
+ (license #5621) patch uploaded by rmudgett JIRA AST-668
+
+2011-09-20 00:56 +0000 [r336877] Russell Bryant <russell at digium.com>
+
+ * res/res_rtp_asterisk.c: Fix crashes in ast_rtcp_write(). This
+ patch addresses crashes related to RTCP handling. The backtraces
+ just show a crash in ast_rtcp_write() where it appears that the
+ RTP instance is no longer valid. There is a race condition with
+ scheduled RTCP transmissions and the destruction of the RTP
+ instance. This patch utilizes the fact that ast_rtp_instance is a
+ reference counted object and ensures that it will not get
+ destroyed while a reference is still around due to scheduled RTCP
+ transmissions. RTCP transmissions are scheduled and executed from
+ the chan_sip scheduler context. This scheduler context is
+ processed in the SIP monitor thread. The destruction of an RTP
+ instance occurs when the associated sip_pvt gets destroyed (which
+ happens when the sip_pvt reference count reaches 0). However, the
+ SIP monitor thread is not the only thread that can cause a
+ sip_pvt to get destroyed. The sip_hangup function, executed from
+ a channel thread, also decrements the reference count on a
+ sip_pvt and could cause it to get destroyed. While this is being
+ changed anyway, the patch also removes calling ast_sched_del()
+ from within the RTCP scheduler callback. It's not helpful. Simply
+ returning 0 prevents the callback from being rescheduled. (closes
+ issue ASTERISK-18570) Related issues that look like they are the
+ same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
+ (issue ASTERISK-15257) (issue ASTERISK-13334) (issue
+ ASTERISK-9977) (issue ASTERISK-9716) Review:
+ https://reviewboard.asterisk.org/r/1444/
+
+2011-09-19 22:07 +0000 [r336791] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_sip.c: Don't interfere with T.38 reinvites This is
+ an update to the fix for ASTERISK-18340 and ASTERISK-17725
+
+2011-09-19 20:27 +0000 [r336733] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * Makefile.rules, include/asterisk/optional_api.h, Makefile,
+ configure, include/asterisk/autoconfig.h.in, main/Makefile,
+ codecs/gsm/Makefile, configure.ac: Various changes to allow 1.8
+ to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6
+ extended to work on 10.7 and later. * Now uses the 'weak' symbol
+ for Lion systems, which no longer support 'weak_import' Closes
+ ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej.
+
+2011-09-19 20:07 +0000 [r336716] Jonathan Rose <jrose at digium.com>
+
+ * res/res_musiconhold.c, apps/app_queue.c, apps/app_mixmonitor.c,
+ apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
+ apps/app_morsecode.c: Document applications that play audio and
+ do not answer unanswered calls. This patch is part of an effort
+ to document early media and its usage. If you are interested in
+ contributing to this documentation effort, there are probably
+ other applications worth documenting as well as an Asterisk wiki
+ article at
+ https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
+
+2011-09-19 18:46 +0000 [r336658] Richard Mudgett <rmudgett at digium.com>
+
+ * UPGRADE.txt, apps/app_dial.c: Made Dial d and H options no longer
+ immediately auto-answer the calling leg. The Dial d and H options
+ break DTMF attended transfer atxferdropcall option. 1) Party A
+ calls party B. 2) Party B does a DTMF attended transfer to Party
+ C. If the dialplan uses the Dial d or H options to call Party C
+ then the Dial application answers the call immediately before
+ initiating the call leg to Party C. The premature answer causes
+ the transfer code to not invoke the atxferdropcall=no behavior
+ for a blonde transfer since Party C has "answered". The transfer
+ code thinks that Party B has "consulted" with Party C when Party
+ B hangs up and completes the transfer to Party A. Party A now
+ hears ringback until Party C actually answers. ASTERISK-13294
+ Dial d option. ASTERISK-11067 Dial H option to disconnect before
+ answer. The referenced issues made Dial answer with the d and H
+ options because many SIP and ISDN phones cannot send DTMF before
+ the call is connected. * Made require the dialplan to control
+ when or if the call needs to be answered to use the Dial
+ application d and H options. (The call is no longer surprise
+ answered when using the Dial d or H options.) Review:
+ https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
+ AST-666
+
+2011-09-19 16:21 +0000 [r336591] Jason Parker <jparker at digium.com>
+
+ * contrib/realtime/postgresql/realtime.sql,
+ configs/cel_odbc.conf.sample, sounds/Makefile,
+ contrib/realtime/mysql/sipfriends.sql,
+ contrib/realtime/mysql/voicemail.sql, cel/cel_odbc.c, /,
+ contrib/realtime/mysql/iaxfriends.sql,
+ contrib/realtime/mysql/meetme.sql: Remove weird mergeinfo props
+ that make merges annoying sometimes.
+
+2011-09-19 15:41 +0000 [r336572] Leif Madsen <lmadsen at digium.com>
+
+ * contrib/scripts/get_ilbc_source.sh: Update get_ilbc_source.sh
+ script to work again. Recently iLBC support in Asterisk has
+ changed after the acquisition of GIPS by Google. More information
+ about how this may affect you is available in a blog post at:
+ http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+
+2011-09-19 15:25 +0000 [r336569] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c: Rework sig_pri_hangup() to be simpler and
+ clearer. JIRA AST-675
+
+2011-09-19 13:33 +0000 [r336501] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Add diversion header to a 302 redirect
+ response if we have diversion data (closes issue ASTERISK-18143)
+ patch by oej
+
+2011-09-19 13:27 +0000 [r336499] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/chan_h323.c: A long time ago in a galaxy far far away a
+ IPv6 update was made, chan_h323 was not updated causeing all to
+ flee to chan_ooh323. the brave Jedi [asterisk developers]
+ pondered this miscarrige of justice and restored order to the
+ force for the sake of closing out 2 old issues. (closes issue
+ ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread,
+ sybasesql Tested by: irroot Reviewed by: IRC (russellb,
+ kpfleming)
+
+2011-09-19 12:06 +0000 [r336378-336440] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c: Make sure manager_debug option is reset at reload
+
+ * Makefile: Revert accidental change that fixes OS/X Lion support
+
+ * Makefile, channels/chan_sip.c: Add missing unlock at MWI message
+ sending time (closes issue ASTERISK-18573) Patches:
+ sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky
+ Thanks to irrot for the reminder, to Gregory for the patch!
+
+2011-09-16 22:10 +0000 [r336312-336314] Terry Wilson <twilson at digium.com>
+
+ * funcs/func_frame_trace.c: Whitespace fix
+
+ * funcs/func_frame_trace.c: Add missing frame types to
+ func_frame_trace Also casts control frames to the proper enum so
+ that the compile will catch new additions.
+
+2011-09-16 19:53 +0000 [r336294] Jonathan Rose <jrose at digium.com>
+
+ * include/asterisk/frame.h, main/channel.c, main/rtp_engine.c,
+ channels/chan_sip.c: Fix bad RTP media bridges in directmedia
+ calls on peers separated by multiple Asterisk nodes. In a
+ situation involving devices on separate Asterisk trunks, the
+ remote RTP bridge would break when starting a call with
+ directmedia. This patch queues a new type of control frame so
+ that our RTP bridge loop can properly detect when these
+ situations occur and check to see if peers need to be updated in
+ order to send their media to the proper location. (Closes issue
+ ASTERISK-18340) Reported by: Thomas Arimont (Closes issue
+ ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose
+
+2011-09-16 19:06 +0000 [r336234] Sean Bright <sean at malleable.com>
+
+ * UPGRADE.txt: Make a note that inotify won't work with an NFS
+ mounted spooler directory.
+
+2011-09-16 10:09 +0000 [r335978-336166] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/chan_misdn.c: The round robin routing routine in
+ chan_misdn.c is broken. it rotates between ports but never checks
+ the channels in the ports. i have extensivly tested it and
+ verified it works on 1 upto 4 ports. before the patch only 1 out
+ of each port was used now all are used as expected. (closes issue
+ ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed
+ by: irroot Review: https://reviewboard.asterisk.org/r/1410/
+
+ * apps/app_queue.c: Locking order in app_queue.c causes deadlocks.
+ a channel lock must never be held with the queues container lock
+ held. the deadlock occured on masquerade. the queues container
+ lock is a relic of the past the old queue module lock. with ao2
+ there is no need to hold this lock when dealing with members this
+ patch removes unneeded locks. (closes issue ASTERISK-18101)
+ (closes issue ASTERISK-18487) Reported by: Paul Rolfe, Jason
+ Legault Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by:
+ Matthew Nicholson Review:
+ https://reviewboard.asterisk.org/r/1402/
+
+ * channels/chan_agent.c: lock the channel before calling
+ ast_bridged_channel() to prevent a seg fault. AMI agents list
+ called on shutdown causes a segfault, introducing proper locking
+ will prevent this. (closes issue ASTERISK-18092) Reported by:
+ agustina Patches: chan_agent.patch (License #5041) patch uploaded
+ by irroot
+
+2011-09-14 18:21 +0000 [r335851-335911] Richard Mudgett <rmudgett at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Remove
+ unnecessary libpri dependency checks in the configure script.
+ Using the --with-pri option with the configure script generated
+ an error about not having PRI_L2_PERSISTENCE if you did not have
+ the absolute latest libpri SVN checkout installed. The
+ AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script
+ seems to be for libraries that are dependent upon other libraries
+ and not necessarily for optional/added features within a library.
+ (closes issue ASTERISK-18535) Reported by: Michael Keuter
+
+ * channels/chan_dahdi.c: Fixed cut-n-paste regression using the
+ wrong variable. Fixes the missing DAHDI channels when using the
+ newer chan_dahdi.conf sections for channel configuration. (closes
+ issue ASTERISK-18496) Reported by: Sean Darcy Patches:
+ jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
+ rmudgett Tested by: Sean Darcy, rmudgett
+
+2011-09-14 13:28 +0000 [r335790] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c: The tech and data members of
+ fast_originate_helper are not string fields. ASTERISK-17709
+
+2011-09-13 22:10 +0000 [r335720] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_directed_pickup.c: Remove obsolete todo comment about
+ PICKUPRESULT.
+
+2011-09-13 21:33 +0000 [r335716] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
+ parse the option "defaultlanguage" from the [options] section of
+ asterisk.conf, as in the sample config file. Otherwise the
+ build-time default language (normally "en") is always the default
+ one. Review: https://reviewboard.asterisk.org/r/1342/
+ Signed-off-by: Tzafrir Cohen (License #5035)
+ <tzafrir.cohen at xorcom.com>
+
+2011-09-13 21:30 +0000 [r335714] Paul Belanger <pabelanger at digium.com>
+
+ * apps/app_meetme.c: Meetme should have 'core' support level
+ (closes issue ASTERISK-18542)
+
+2011-09-13 18:52 +0000 [r335655] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * configure, configure.ac: Move mandatory checks closer to the
+ beginning of the file. If these are going to fail, they should
+ fail as quickly as possible.
+
+2011-09-13 18:20 +0000 [r335618] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, main/manager.c: Don't limit the size of appdata for
+ manager originate actions. ASTERISK-17709 Patch by: tilghman
+ (with modifications)
+
+2011-09-13 07:11 +0000 [r335497] Russell Bryant <russell at digium.com>
+
+ * main/event.c, include/asterisk/event.h, res/ais/evt.c: Fix a
+ crash in res_ais. This patch resolves a crash observed in a load
+ testing environment that involved the use of the res_ais module.
+ I observed some crashes where the event delivery callback would
+ get called, but the length parameter incidcating how much data
+ there was to read was 0. The code assumed (with good reason I
+ would think) that if this callback got called, there was an event
+ available to read. However, if the rare case that there's nothing
+ there, catch it and return instead of blowing up. More
+ specifically, the change always ensure that the size of the
+ received event in the cluster is always big enough to be a real
+ ast_event. Review: https://reviewboard.asterisk.org/r/1423/
+
+2011-09-12 15:54 +0000 [r335431-335433] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c: Properly set caller_warning and callee_warning
+ before we try to use them. ASTERISK-18199 Patch by: elguero
+ Testing by: rtang
+
+ * bridges/bridge_multiplexed.c: Prevent a race condition when the
+ bridge technology changes. This change was ported from asterisk
+ 10. ASTERISK-18155
+
+2011-09-12 14:21 +0000 [r335320-335341] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_dial.c: Ensure frames are not written to dialed channel
+ if ringback is requested When a single channel was dialed and
+ there was media to be forwarded to the calling channel, the media
+ was written without regard for ringback causing silence to be
+ heard in some circumstances. This regression was introduced when
+ the meaning of "single" changed to mean only the number of
+ channels dialed. (closes issue ASTERISK-18083)
+
+ * channels/chan_iax2.c: Prevent IAX2 from getting IPv6 addresses
+ via DNS IAX2 does not support IPv6 and getting such addresses
+ from DNS can cause error messages on the remote end involving bad
+ IPv4 address casts in the presence of IPv6/IPv4 tunnels. This
+ patch ensures that IAX2 will not encounter IPv6 addresses via DNS
+ queries. (closes issue ASTERISK-18090)
+
+2011-09-12 13:25 +0000 [r335319] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Lock the peer->mvipvt to avoid crashes with
+ SIP history enabled After the launch of 1.6 event-based MWI we
+ have two threads handling the peer->mwipvt, which cause issues
+ with SIP history additions in combination with the max limit for
+ number of history entries. Review:
+ https://reviewboard.asterisk.org/r/1373/ (closes issue
+ ASTERISK-18288) Thanks to irrot for peer review. Work with this
+ bug funded by IPvision AS
+
+2011-09-12 11:09 +0000 [r335259] Stefan Schmidt <sst at sil.at>
+
+ * channels/chan_sip.c: build_peer doesnt unlink a peer object from
+ peers_by_ip container which leads to a wrong refcounter value.
+ adding an ao2_unlink from the peers_by_ip container fix it.
+ Review: https://reviewboard.asterisk.org/r/1428/
+
+2011-09-09 16:09 +0000 [r335064] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
+ main/channel.c, channels/chan_usbradio.c, main/dial.c,
+ channels/chan_dahdi.c, channels/chan_misdn.c,
+ channels/chan_skinny.c, funcs/func_frame_trace.c,
+ main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
+ include/asterisk/frame.h, channels/sig_ss7.c,
+ channels/chan_mgcp.c, apps/app_dial.c, channels/chan_unistim.c,
+ main/pbx.c, addons/chan_ooh323.c, channels/chan_sip.c: Updated
+ SIP 484 handling; added Incomplete control frame When a SIP phone
+ uses the dial application and receives a 484 Address Incomplete
+ response, if overlapped dialing is enabled for SIP, then the 484
+ Address Incomplete is forwarded back to the SIP phone and the
+ HANGUPCAUSE channel variable is set to 28. Previously, the
+ Incomplete application dialplan logic was automatically
+ triggered; now, explicit dialplan usage of the application is
+ required. Additionally, this patch adds a new AST_CONTOL_FRAME
+ type called AST_CONTROL_INCOMPLETE. If a channel driver receives
+ this control frame, it is an indication that the dialplan expects
+ more digits back from the device. If the device supports overlap
+ dialing it should attempt to notify the device that the dialplan
+ is waiting for more digits; otherwise, it can handle the frame in
+ a manner appropriate to the channel driver. (closes issue
+ ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew
+ Jordan Review: https://reviewboard.asterisk.org/r/1416/
+
+2011-09-08 22:27 +0000 [r334953] Richard Mudgett <rmudgett at digium.com>
+
+ * main/logger.c: Fix crash with res_fax when MALLOC_DEBUG and "core
+ stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
+ enabled when res_fax tries to unregister its logger level. * Make
+ ast_logger_unregister_level() use ast_free() instead of free().
+ When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
+ a call to free(). Therefore, if you allocated memory with a form
+ of ast_malloc you must free it with ast_free.
+
+2011-09-07 19:35 +0000 [r334843] Paul Belanger <pabelanger at digium.com>
+
+ * channels/chan_iax2.c: Cleanup chan_iax2.c log messages Review:
+ https://code.asterisk.org/code/cru/CR-AST-11
+
+2011-09-07 19:31 +0000 [r334840] Richard Mudgett <rmudgett at digium.com>
+
+ * main/features.c: Fix AMI action Park crash. * Made AMI action
+ Park not say anything to the parker channel (AMI header Channel2)
+ since the AMI action is a third party parking the call. (This is
+ a change in behavior that cannot be preserved without a lot of
+ effort.) * Made not play pbx-parkingfailed if the Park 's' option
+ is used. JIRA AST-660
+
+2011-09-07 13:26 +0000 [r334682] Stefan Schmidt <sst at sil.at>
+
+ * main/features.c: Adding the Feature to sent a Reason Header in a
+ SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
+ before doing a masquerade in the pickup function.
+
+2011-09-07 08:12 +0000 [r334616-334620] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * CHANGES, apps/app_queue.c: peroid typo
+
+ * main/pbx.c: Prevent segfault if call arrives before Asterisk is
+ fully booted. Prevent ast_pbx_start and ast_run_start from
+ starting a new thread unless asterisk is fully booted. alecdavis
+ (license 585) Tested by: alecdavis Review:
+ https://reviewboard.asterisk.org/r/1407/
+
+2011-09-06 13:48 +0000 [r334453] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * apps/app_voicemail.c: Make SQL query in app_voicemail.c portable
+ LIMIT is not portable. Regression from r312212 (closes issue
+ ASTERISK-18255) Reported by: Leif Madsen Tested by: Leif Madsen
+ Review: https://reviewboard.asterisk.org/r/1415/
+
+2011-09-23 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.7.0 Released.
+
+2011-09-19 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.7.0-rc2 Released.
+
+ * r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) |
+ 11 lines
+
+ Fixed cut-n-paste regression using the wrong variable.
+
+ Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
+ sections for channel configuration.
+
+ (closes issue ASTERISK-18496)
+
+ * r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) |
+ 13 lines
+
+ Remove unnecessary libpri dependency checks in the configure script.
+
+ Using the --with-pri option with the configure script generated an
+ error
+ about not having PRI_L2_PERSISTENCE if you did not have the absolute
+ latest libpri SVN checkout installed.
+
+ The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems
+ to
+ be for libraries that are dependent upon other libraries and not
+ necessarily for optional/added features within a library.
+
+ (closes issue ASTERISK-18535)
+
+ * r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) | 7
+ lines
+
+ Update get_ilbc_source.sh script to work again.
+
+ Recently iLBC support in Asterisk has changed after the acquisition of
+ GIPS
+ by Google. More information about how this may affect you is available
+ in a
+ blog post at:
+
+ http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
+
+ * r335714 | pabelanger | 2011-09-13 16:30:18 -0500 (Tue, 13 Sep 2011)
+ | 4 lines
+
+ Meetme should have 'core' support level
+
+ (closes issue ASTERISK-18542)
+
[... 33741 lines stripped ...]
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