[asterisk-commits] bebuild: tag 10.0.0-beta2 r338080 - /tags/10.0.0-beta2/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Sep 27 12:40:13 CDT 2011
Author: bebuild
Date: Tue Sep 27 12:40:09 2011
New Revision: 338080
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=338080
Log:
Importing files for 10.0.0-beta2 release.
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tags/10.0.0-beta2/.version (with props)
tags/10.0.0-beta2/ChangeLog (with props)
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+2011-09-27 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.0.0-beta2 Released.
+
+ * Based on revision that passed automated testing
+ (http://bamboo.asterisk.org/browse/AST10-LUCID-178)
+
+2011-09-26 19:35 +0000 [r337974] Richard Mudgett <rmudgett at digium.com>
+
+ * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c,
+ apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
+ include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
+ 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: Merged revisions 337973
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
+ | 30 lines 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 19:18 +0000 [r337840-337902] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337898
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) |
+ 4 lines Spelling fix ........
+
+ * /, apps/app_queue.c: Merged revisions 337839 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) |
+ 11 lines 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 00:45 +0000 [r337775] Russell Bryant <russell at digium.com>
+
+ * configs/res_pktccops.conf.sample, /: Merged revisions 337774 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011)
+ | 11 lines 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 21:37 +0000 [r337721] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 337720 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011)
+ | 18 lines 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 18:43 +0000 [r337640] Paul Belanger <pabelanger at digium.com>
+
+ * CREDITS, apps/app_meetme.c, CHANGES: Revert previous commit New
+ feature should be added into trunk, unfortunately it is too late
+ for the Asterisk 10 branch.
+
+2011-09-22 15:47 +0000 [r337595-337597] Jonathan Rose <jrose at digium.com>
+
+ * channels/sip/security_events.c (added),
+ channels/sip/include/security_events.h (added): Forgot to svn add
+ new files to r337595 Part of Generating security events for
+ chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young
+ Patches: security_events_chan_sip_v4.patch (License #5026) by
+ Michael L. Young Reviewboard:
+ https://reviewboard.asterisk.org/r/1362/
+
+ * configs/logger.conf.sample, channels/chan_sip.c,
+ include/asterisk/event_defs.h, main/security_events.c,
+ main/event.c, CHANGES, channels/sip/include/sip.h,
+ include/asterisk/security_events_defs.h: Generate Security events
+ in chan_sip using new Security Events Framework Security Events
+ Framework was added in 1.8 and support was added for AMI to
+ generate events at that time. This patch adds support for
+ chan_sip to generate security events. (closes issue
+ ASTERISK-18264) Reported by: Michael L. Young Patches:
+ security_events_chan_sip_v4.patch (license #5026) by Michael L.
+ Young Review: https://reviewboard.asterisk.org/r/1362/
+
+2011-09-22 11:44 +0000 [r337431-337542] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * res/res_srtp.c, /: Merged revisions 337541 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) |
+ 8 lines 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: Merged revisions 337486 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22
+ Sep 2011) | 10 lines 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)
+ ........
+
+ * apps/app_originate.c, CHANGES: Revert commit r337261 This commit
+ is for trunk not version 10 ----- Adds a timeout argument to
+ app_originate the default is 30s this will be used if the timout
+ supplied is invalid or no timeout is supplied. -----
+
+ * main/channel.c, /: Merged revisions 337430 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) |
+ 19 lines 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 21:25 +0000 [r337342-337380] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * apps/app_voicemail.c, /: More silly spacing changes ..... Merged
+ revisions 337353 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: ........ Dumb little spacing fix.
+ ........ Merged revisions 337344 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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 ........ Merged revisions
+ 337325 from https://origsvn.digium.com/svn/asterisk/branches/1.8
+
+2011-09-21 11:15 +0000 [r337261-337263] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * configs/sip.conf.sample: Whitespace fixup from SRTP patch
+
+ * apps/app_originate.c, CHANGES: Adds a timeout argument to
+ app_originate the default is 30s this will be used if the timout
+ supplied is invalid or no timeout is supplied. Contributed by:
+ jacco (thank you for the work) Review:
+ https://reviewboard.asterisk.org/r/1310/
+
+2011-09-21 09:32 +0000 [r337178-337219] Olle Johansson <oej at edvina.net>
+
+ * configs/extensions.conf.sample, main/pbx.c, CHANGES: Make
+ ast_pbx_run() not default to s at default if extension is not found
+ Review: https://reviewboard.asterisk.org/r/1446/ This is a bug -
+ or architecture mistake - that has been in Asterisk for a very
+ long time. It was exposed by the AMI originate action and
+ possibly some other applications. Most channel drivers checks if
+ an extension exists BEFORE starting a pbx on an inbound call, so
+ most calls will not depend on this issue. Thanks everyone
+ involved in the review and on IRC and the mailing list for a
+ quick review and all the feedback. (closes issue ASTERISK-18578)
+
+ * res/res_rtp_asterisk.c, configs/rtp.conf.sample, CHANGES: Change
+ strictrtp option to default to yes in the RTP module Suggested by
+ Kapejod on Facebook Review:
+ https://reviewboard.asterisk.org/r/1448/ (closes issue
+ ASTERISK-18587) Thanks for quick feedback to kpfleming and
+ Tilghman --Denna och nedanstående rader kommer inte med i
+ loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M
+ res/res_rtp_asterisk.c
+
+2011-09-20 22:49 +0000 [r337120] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_voicemail.c, apps/app_dial.c, include/asterisk/app.h, /,
+ apps/app_meetme.c, apps/app_minivm.c, main/app.c,
+ apps/app_confbridge.c, apps/app_followme.c: Merged revisions
+ 337118 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011)
+ | 21 lines 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 22:47 +0000 [r337119] Richard Mudgett <rmudgett at digium.com>
+
+ * funcs/func_strings.c: Fix crash with STRREPLACE function. The
+ ast_func_read() function calls the .read2 callback with the len
+ parameter set to zero indicating no size restrictions on the
+ supplied ast_str buffer. The value was used to dimension a local
+ starts[] array with the array subsequently used. * Reworked the
+ strreplace() function to perform the string replacement in a
+ straight forward manner. Eliminated the need for the starts[]
+ array. (closes issue ASTERISK-18545) Reported by: Federico Alves
+ Patches: jira_asterisk_18545_v10.patch (license #5621) patch
+ uploaded by rmudgett Tested by: rmudgett, Federico Alves
+
+2011-09-20 22:19 +0000 [r337116] Leif Madsen <lmadsen at digium.com>
+
+ * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011)
+ | 7 lines 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 21:05 +0000 [r337062] Kinsey Moore <kmoore at digium.com>
+
+ * tests/test_pbx.c, main/pbx.c, /: Merged revisions 337061 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) |
+ 11 lines 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 19:12 +0000 [r336978-337008] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_ss7.c: Merged revisions 337007 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011)
+ | 15 lines 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: Merged revisions 336977 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011)
+ | 21 lines 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 16:51 +0000 [r336936] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/sip/sdp_crypto.c, channels/chan_sip.c,
+ channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h,
+ configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h:
+ Allow Setting Auth Tag Bit length Based on invite or config
+ option Update the SIP SRTP API to allow use of 32 or 80 bit
+ taglen. Curently only 80 bit is supported. The outgoing invite
+ will use the taglen of the incoming invite preventing one-way
+ audio. (Closes issue ASTERISK-17895) Review:
+ https://reviewboard.asterisk.org/r/1173/
+
+2011-09-20 01:03 +0000 [r336878] Russell Bryant <russell at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Merged revisions 336877 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19
+ Sep 2011) | 36 lines 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 22:13 +0000 [r336792] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 336791 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011)
+ | 2 lines Don't interfere with T.38 reinvites This is an update
+ to the fix for ASTERISK-18340 and ASTERISK-17725 ........
+
+2011-09-19 21:41 +0000 [r336734-336789] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * funcs/func_strings.c: Ensure substring will not be found in the
+ previous match.
+
+ * include/asterisk/optional_api.h, Makefile, /, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile,
+ codecs/gsm/Makefile, configure.ac, Makefile.rules: Merged
+ revisions 336733 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011)
+ | 11 lines 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 20:16 +0000 [r336717] Jonathan Rose <jrose at digium.com>
+
+ * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
+ apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c,
+ apps/app_mixmonitor.c: Merged revisions 336716 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) |
+ 7 lines 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 18:51 +0000 [r336659] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336658 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011)
+ | 31 lines 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 15:42 +0000 [r336573] Leif Madsen <lmadsen at digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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/
+ ........
+
+2011-09-19 15:32 +0000 [r336570] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/sig_pri.c: Merged revisions 336569 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011)
+ | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA
+ AST-675 ........
+
+2011-09-19 13:48 +0000 [r336502-336504] Olle Johansson <oej at edvina.net>
+
+ * Makefile: Revert accidental change
+
+ * Makefile, /, channels/chan_sip.c: Merged revisions 336501 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336501 | oej | 2011-09-19 15:33:50 +0200 (MÃ¥n, 19 Sep 2011) | 5
+ lines Add diversion header to a 302 redirect response if we have
+ diversion data (closes issue ASTERISK-18143) patch by oej
+ ........
+
+2011-09-19 13:31 +0000 [r336500] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * /, channels/chan_h323.c: Merged revisions 336499 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19
+ Sep 2011) | 13 lines 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 12:15 +0000 [r336381-336441] Olle Johansson <oej at edvina.net>
+
+ * main/manager.c, /: Merged revisions 336440 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336440 | oej | 2011-09-19 14:06:48 +0200 (MÃ¥n, 19 Sep 2011) | 2
+ lines Make sure manager_debug option is reset at reload ........
+
+ * /, channels/chan_sip.c: Merged revisions 336378 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336378 | oej | 2011-09-19 11:40:44 +0200 (MÃ¥n, 19 Sep 2011) | 9
+ lines 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 22:11 +0000 [r336313-336316] Terry Wilson <twilson at digium.com>
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336314 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16
+ Sep 2011) | 2 lines Whitespace fix ........
+
+ * /, funcs/func_frame_trace.c: Merged revisions 336312 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16
+ Sep 2011) | 5 lines 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 21:09 +0000 [r336307] Jonathan Rose <jrose at digium.com>
+
+ * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c,
+ include/asterisk/frame.h: Merged revisions 336294 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep
+ 2011) | 13 lines 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 19:10 +0000 [r336235] Sean Bright <sean at malleable.com>
+
+ * /, UPGRADE-1.8.txt: Merged revisions 336234 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, 16 Sep
+ 2011) | 2 lines Make a note that inotify won't work with an NFS
+ mounted spooler directory. ........
+
+2011-09-16 10:12 +0000 [r336094-336167] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * channels/chan_misdn.c, /: Merged revisions 336166 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16
+ Sep 2011) | 16 lines 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: Merged revisions 336093 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) |
+ 20 lines 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/
+ ........
+
+2011-09-15 15:19 +0000 [r336091] David Vossel <dvossel at digium.com>
+
+ * main/format_cap.c: Removes some no-op code found in format_cap.c.
+
+2011-09-15 12:46 +0000 [r336042] Olle Johansson <oej at edvina.net>
+
+ * CREDITS, apps/app_meetme.c, CHANGES: Meetme: Introducing a new
+ option "k" to kill a conference if there's only a single member
+ left. When using Meetme as a modular call bridge from third party
+ applications, it's handy to make it behave like a normal call
+ bridge. When the second to last person exists, the last person
+ will be kicked out of the conference when this option is enabled.
+ (closes issue ASTERISK-18234) Review:
+ https://reviewboard.asterisk.org/r/1376/ Patch by oej, sponsored
+ by ClearIT, Solna, Sweden
+
+2011-09-15 08:29 +0000 [r335991] Gregory Nietsky <gregory at distrotech.co.za>
+
+ * /, channels/chan_agent.c: Merged revisions 335978 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15
+ Sep 2011) | 11 lines 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 18:31 +0000 [r335852-335912] Richard Mudgett <rmudgett at digium.com>
+
+ * /, configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Merged revisions 335911 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ 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) Reported
+ by: Michael Keuter ........
+
+ * channels/chan_dahdi.c, /: Merged revisions 335851 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ 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) 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 [r335791] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/manager.c, /: Merged revisions 335790 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep
+ 2011) | 4 lines The tech and data members of
+ fast_originate_helper are not string fields. ASTERISK-17709
+ ........
+
+2011-09-13 22:10 +0000 [r335721] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_directed_pickup.c: Merged revisions 335720 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 Sep 2011)
+ | 1 line Remove obsolete todo comment about PICKUPRESULT.
+ ........
+
+2011-09-13 21:37 +0000 [r335717] 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> Original-Commit:
+ http://svn.digium.com/svn/asterisk/branches/1.8@335716
+
+2011-09-13 18:55 +0000 [r335656] Tilghman Lesher <tilghman at meg.abyt.es>
+
+ * /, configure, configure.ac: Merged revisions 335655 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13
+ Sep 2011) | 4 lines 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 18:47 +0000 [r335653] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/pbx.c, main/manager.c, /: Merged revisions 335618 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep
+ 2011) | 5 lines Don't limit the size of appdata for manager
+ originate actions. ASTERISK-17709 Patch by: tilghman (with
+ modifications) ........
+
+2011-09-13 07:24 +0000 [r335510] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged
+ revisions 335497 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011)
+ | 15 lines 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 15:55 +0000 [r335434] Matthew Nicholson <mnicholson at digium.com>
+
+ * main/channel.c, /: Merged revisions 335433 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep
+ 2011) | 6 lines Properly set caller_warning and callee_warning
+ before we try to use them. ASTERISK-18199 Patch by: elguero
+ Testing by: rtang ........
+
+2011-09-12 14:22 +0000 [r335346] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_dial.c, /: Merged revisions 335341 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) |
+ 10 lines 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) ........
+
+2011-09-12 13:47 +0000 [r335323] Olle Johansson <oej at edvina.net>
+
+ * /, channels/chan_sip.c: Merged revisions 335319 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+ r335319 | oej | 2011-09-12 15:25:30 +0200 (MÃ¥n, 12 Sep 2011) | 12
+ lines 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 13:27 +0000 [r335321] Kinsey Moore <kmoore at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 335320 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.8
+ ........ r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12
+ Sep 2011) | 9 lines 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) ........
+
[... 16948 lines stripped ...]
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