[svn-commits] bebuild: tag 1.8.28.0-rc1 r414408 - /tags/1.8.28.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu May 22 11:15:31 CDT 2014
Author: bebuild
Date: Thu May 22 11:15:27 2014
New Revision: 414408
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=414408
Log:
Importing files for 1.8.28.0-rc1 release.
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+2014-05-22 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.28.0-rc1 Released.
+
+2014-05-22 15:47 +0000 [r414401] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_meetme.c: app_meetme: Don't interrupt MOH for waitmarked
+ users. Occasionally, when the last marked user leaves the
+ conference, waitmarked users don't get MOH if MOH is supposed to
+ be played while a waitmarked user is waiting for another marked
+ user. * Made not interrupt MOH when the user is a waitmarked
+ user. The waitmarked user doesn't need to hear any leave
+ announcements from the conference as the user would have already
+ heard different leave announcements if they were enabled.
+ Apparently DAHDI occasionally sends unending non-silent streams
+ to these users or a normal user still in the conference has
+ continuous high background noise. These non-silent streams cause
+ MOH to be suspended while the never ending "announcement" is
+ played. Issue caused by ASTERISK-13680. AST-1349 #close Reported
+ by: Tyler Stewart Review:
+ https://reviewboard.asterisk.org/r/3543/
+
+2014-05-22 13:58 +0000 [r414345] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt: UPGRADE: Add note for REF_DEBUG flag
+
+2014-05-21 22:01 +0000 [r414269] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_local.c: chan_local: Only block media frames when a
+ generator is on both ends of a local channel. The fix for
+ ASTERISK-12292 was a bit too aggressive. You could have
+ generators pointed at each other on local channels but need to
+ get other kinds of frames such as DTMF or CONNECTED_LINE frames
+ accross.
+
+2014-05-21 18:58 +0000 [r414214] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * funcs/func_strings.c: pbx.c: prevent potential crash from
+ recursive replace() Recurisve usage of replace() resulted in
+ corruption of the temporary string storage and potential crash.
+ By changing the string to be allocated separtely per instance,
+ this is eliminated. ASTERISK-23650 #comment Reported by: Roel van
+ Meer ASTERISK-23650 #close Review:
+ https://reviewboard.asterisk.org/r/3539/
+
+2014-05-19 13:31 +0000 [r414152] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: chan_ooh323: fix h323_log full path name *
+ fix to use astlogdir option for h323_log file instead of
+ hardcoded ASTERISK-23754 #close Reported by: Igor Goncharovsky
+ Patches: ooh323_logger_patch.diff
+
+2014-05-16 20:00 +0000 [r413991-414067] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_dahdi.c: chan_dahdi: Fix analog dialtone detection.
+ * Check if waitingfordt (waitfordialtone) is enabled in
+ dahdi_read() to allow the DSP to operate early enough to detect
+ dialtone. * Made use the correct variable in
+ my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve
+ Davies Patches: dialtone_detect_fix (license #5012) patch
+ uploaded by Steve Davies Review:
+ https://reviewboard.asterisk.org/r/3534/
+
+ * apps/app_meetme.c: app_meetme: Fix overwrite of DAHDI conference
+ data structure. Starting a conference recording using the admin
+ menu overwrites the DAHDI conference data structure used to
+ modify the admin user's conference mute mode. * Made no longer
+ pass the user's DAHDI conference data structure into the menu
+ functions. The menu now uses its own DAHDI conference data
+ structure to start the recording channel. * Moved the unlock
+ conf->playlock to before playing the conf-full message. No sense
+ keeping the lock while that prompt is playing. The user is never
+ going to get into the conference at that point.
+
+2014-05-15 15:32 +0000 [r413949] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * apps/app_dial.c, channels/chan_local.c, UPGRADE.txt:
+ chan_local+app_dial: Propagagate call answered elsewhere over
+ local channels. AST_FLAG_ANSWERED_ELSEWHERE was not propagated
+ back from local channels. It is now. That means that when a call
+ is picked up from a callgroup of local channels, the other
+ channels will now properly see it as "picked up". This occurs
+ when you use a construct like
+ Dial(Local/a at context&Local/b at context) where a at context and
+ b at context dial two chan_sip devices respectively. If one device
+ picks up, the other will not see "1 missed call" anymore. In this
+ respect, it now behaves the same as when doing Dial(SIP/a&SIP/b).
+ Review: https://reviewboard.asterisk.org/r/3540/
+
+2014-05-14 15:27 +0000 [r413894] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * res/res_musiconhold.c: res_musiconhold: Minor cleanup. Fix a few
+ free()'s that should be ast_free()'s. Reverted an old workaround
+ that isn't necessary. Reorder a tiny bit of code. Remove a bit of
+ commented-out code. Review:
+ https://reviewboard.asterisk.org/r/3536/
+
+2014-05-13 14:32 +0000 [r413787-413832] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * channels/chan_sip.c: chan_sip+CEL: Add missing ANSWER and PICKUP
+ events to INVITE/w/replaces pickup. When doing a "BLF-style call
+ pickup" -- an INVITE with Replaces: header -- the CEL log would
+ lack the ANSWER and PICKUP events. This patch adds the two
+ missing events to the handle_invite_replaces() function.
+ ASTERISK-22977 #close Review:
+ https://reviewboard.asterisk.org/r/3073/
+
+ * main/rtp_engine.c: rtp: Fix case typo in H263+ mime.
+ http://tools.ietf.org/html/rfc3555#section-4.2.6 says the
+ canonical mime subtype is "H263-1998", not "h263-1998". Original
+ code was added in r183101 on 2009-03-19 02:26:50 +0100. This
+ fixes issues with Polycom phones. ASTERISK-23665 #close
+ ASTERISK-23665 #comment Patch r3529.patch uploaded by Guillaume
+ Maudoux, backported by me. Review:
+ https://reviewboard.asterisk.org/r/3529/
+
+2014-05-12 23:08 +0000 [r413714] Richard Mudgett <rmudgett at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ channels/sig_pri.c: chan_dahdi/sig_pri: Prevent unnecessary
+ PROGRESS events when overlap dialing is enabled. When overlap
+ dialing is enabled, the lack of inband audio available
+ information in the SETUP_ACKNOWLEDGE events causes an
+ interoperability problem with SIP. sig_pri doesn't know if there
+ is dialtone present when a SETUP_ACKNOWLEDGE is received so it
+ assumes it is there and posts an AST_CONTROL_PROGRESS frame. The
+ SIP channel driver then sends out a 183 Session Progress and
+ blocks the desired 180 Ringing message when the ALERTING message
+ comes in. * Made the configure script detect if the installed
+ version of libpri supports the SETUP_ACKNOWLEDGE enhancements. *
+ Using the new API, made generate an AST_CONTROL_PROGRESS frame on
+ an incoming SETUP_ACKNOWLEDGE message when the message indicates
+ inband audio is present instead of assuming that dialtone is
+ present. * Using the new API, made SETUP_ACKNOWLEDGE send out an
+ inband audio available indication only if dialtone is expected.
+ The change also makes the fallback behaviour of sending the
+ PROGRESS message better by sending it only if dialtone is
+ expected. * Changed receiving a PROCEEDING message to not
+ generate an AST_CONTROL_PROGRESS frame if the progress indication
+ ie indicates non-end-to-end-ISDN. This helps interoperability
+ with SIP. * Changed sending a PROCEEDING message in response to
+ an AST_CONTROL_PROCEEDING frame to not indicate inband audio
+ available. It was silly to do so anyway because the channel
+ driver doesn't know if inband audio is even available. This helps
+ interoperability with SIP. This patch and a corresponding change
+ in libpri work together to allow Asterisk to control the inband
+ audio available progress indication ie on the SETUP_ACKNOWLEDGE
+ message when dialtone is present. AST-1338 #close Reported by:
+ Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/
+
+2014-05-09 23:02 +0000 [r413586-413592] Kinsey Moore <kmoore at digium.com>
+
+ * funcs/func_env.c: Fix 32bit build for func_env
+
+ * channels/chan_sip.c: Fix 32bit build for chan_sip
+
+ * channels/chan_dahdi.c, channels/sig_analog.c,
+ include/asterisk/astobj.h, main/event.c, funcs/func_iconv.c,
+ channels/sip/config_parser.c, apps/app_stack.c, res/res_odbc.c,
+ apps/app_adsiprog.c, res/res_calendar.c, main/udptl.c,
+ main/stun.c, main/frame.c, channels/chan_sip.c,
+ apps/app_festival.c, funcs/func_env.c, main/taskprocessor.c,
+ channels/chan_iax2.c, apps/app_getcpeid.c, res/res_monitor.c,
+ res/ael/pval.c, main/channel.c, main/manager.c,
+ formats/format_pcm.c, funcs/func_srv.c, main/file.c,
+ main/callerid.c, main/app.c, channels/chan_alsa.c, main/adsi.c,
+ pbx/pbx_dundi.c, main/stdtime/localtime.c, res/res_fax_spandsp.c,
+ main/sched.c, res/res_rtp_asterisk.c, cel/cel_pgsql.c,
+ cdr/cdr_adaptive_odbc.c, res/res_musiconhold.c,
+ channels/chan_gtalk.c, channels/sig_pri.c, res/res_srtp.c,
+ main/io.c, channels/chan_jingle.c, channels/chan_phone.c,
+ funcs/func_enum.c, res/res_config_odbc.c, apps/app_minivm.c,
+ res/res_agi.c, main/features.c, apps/app_dumpchan.c,
+ main/abstract_jb.c, main/logger.c, apps/app_sms.c,
+ main/audiohook.c, pbx/pbx_config.c, main/bridging.c, main/dsp.c,
+ apps/app_voicemail.c, apps/app_dial.c,
+ res/res_calendar_exchange.c, main/security_events.c,
+ res/res_fax.c, res/res_timing_dahdi.c, funcs/func_sysinfo.c,
+ main/utils.c, main/devicestate.c, res/res_jabber.c,
+ res/res_pktccops.c, main/cli.c, main/data.c, cel/cel_odbc.c,
+ channels/chan_skinny.c, main/asterisk.c,
+ channels/sip/include/sip.h, channels/chan_mgcp.c, main/xmldoc.c,
+ channels/chan_unistim.c, main/pbx.c,
+ res/res_calendar_icalendar.c, channels/chan_local.c,
+ main/rtp_engine.c, main/ccss.c, main/translate.c,
+ res/res_crypto.c, res/res_calendar_caldav.c, main/aoc.c,
+ pbx/dundi-parser.c, main/cel.c, apps/app_queue.c, main/enum.c,
+ channels/iax2-parser.c, main/config.c, res/res_calendar_ews.c,
+ main/netsock.c, main/loader.c: Allow Asterisk to compile under
+ GCC 4.10 This resolves a large number of compiler warnings from
+ GCC 4.10 which cause the build to fail under dev mode. The vast
+ majority are signed/unsigned mismatches in printf-style format
+ strings.
+
+2014-05-08 00:33 +0000 [r413485] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_queue.c, main/manager.c: app_queue: Extend documentation
+ for various Manager actions and events.
+
+2014-05-07 17:46 +0000 [r413396] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_config_odbc.c: Fix encoding of custom prepare extra data.
+ Patches: res_config_odbc-take2.patch by John Hardin (License
+ #6512)
+
+2014-05-06 16:57 +0000 [r413304] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_config_odbc.c: Ensure that all parts of SQL UPDATEs and
+ DELETEs are encoded. Patches: res_config_odbc.patch by John
+ Hardin (License #6512)
+
+2014-05-02 20:21 +0000 [r413224-413241] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_config_odbc.c: Prevent crashes in res_config_odbc due to
+ uninitialized string fields. Patches: odbc-crash.patch by John
+ Hardin (License #6512)
+
+ * res/res_config_pgsql.c: Return the number of rows affected by a
+ SQL insert, rather than an object ID. The realtime API specifies
+ that the store callback is supposed to return the number of rows
+ affected. res_config_pgsql was instead returning an Oid cast as
+ an int, which during any nominal execution would be cast to 0.
+ Returning 0 when more than 0 rows were inserted causes problems
+ to the function's callers. To give an idea of how strange code
+ can be, this is the necessary code change to fix a device state
+ issue reported against chan_pjsip in Asterisk 12+. The issue was
+ that the registrar would attempt to insert contacts into the
+ database. Because of the 0 return from res_config_pgsql, the
+ registrar would think that the contact was not successfully
+ inserted, even though it actually was. As such, even though the
+ contact was query-able and it was possible to call the endpoint,
+ Asterisk would "think" the endpoint was unregistered, meaning it
+ would report the device state as UNAVAILABLE instead of
+ NOT_INUSE. The necessary fix applies to all versions of Asterisk,
+ so even though the bug reported only applies to Asterisk 12+, the
+ code correction is being inserted into 1.8+. Closes issue
+ ASTERISK-23707 Reported by Mark Michelson
+
+2014-04-23 17:47 +0000 [r412922] Richard Mudgett <rmudgett at digium.com>
+
+ * main/http.c: http: Fix spurious ERROR message in responses with
+ no content. Backport -r411687 and fix the fix because
+ content_length is the length of out plus the length of the file
+ controlled by fd. When a response has an out content length of 0,
+ fwrite would be called to write a buffer with no data in it. This
+ resulted in the following classic error message: [Apr 3 11:49:17]
+ ERROR[26421] http.c: fwrite() failed: Success This patch makes it
+ so that we only attempt to write the content of out if the out
+ string is non-zero.
+
+2014-04-21 17:51 +0000 [r412764-412821] Jonathan Rose <jrose at digium.com>
+
+ * CHANGES: chan_sip: trust_id_outbound CHANGES message improvement
+ (closes issue AST-1301) (closes issue ASTERISK-19465) Reported
+ by: Krzysztof Chmielewski
+
+ * CHANGES: Typo in CHANGES
+
+2014-04-21 15:50 +0000 [r412745] Kinsey Moore <kmoore at digium.com>
+
+ * main/manager.c, main/http.c: HTTP: Add TCP_NODELAY to accepted
+ connections This adds the TCP_NODELAY option to accepted
+ connections on the HTTP server built into Asterisk. This option
+ disables the Nagle algorithm which controls queueing of outbound
+ data and in some cases can cause delays on receipt of response by
+ the client due to how the Nagle algorithm interacts with TCP
+ delayed ACK. This option is already set on all non-HTTP AMI
+ connections and this change would cover standard HTTP requests,
+ manager HTTP connections, and ARI HTTP requests and websockets in
+ Asterisk 12+ along with any future use of the HTTP server.
+ Review: https://reviewboard.asterisk.org/r/3466/
+
+2014-04-21 15:25 +0000 [r412744] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
+ channels/sip/include/sip.h: chan_sip: Add sendrpid trust options
+ In r411189, some behavior was changed which made sendrpid
+ behavior act in a more trusting manner by sending full user data
+ for peers set with private caller presence in P-Asserted-Identity
+ headers. Since this changed long time expected behaviors, we
+ decided to pull that patch when that was pointed out by the
+ community. Instead, this patch provides a trust_id_outbound
+ setting which will expose the data per RFC-3325 if set to 'yes'
+ and simply not send the PAI/RPID headers at all if set to 'no'.
+ By default trust_id_outbound will be set to 'legacy' which will
+ preserve the behavior prior to these patches. Extra special
+ thanks to Walter Doekes for providing advice and feedback.
+ (closes issue AST-1301) (closes issue ASTERISK-19465) Reported
+ by: Krzysztof Chmielewski Review:
+ https://reviewboard.asterisk.org/r/3447/
+
+2014-04-19 01:01 +0000 [r412655] Matthew Jordan <mjordan at digium.com>
+
+ * apps/app_sms.c: app_sms: Fix uninitialized values; hangup channel
+ when REL is sent successfully This patch fixes two issues in
+ app_sms: (1) Firstly, the 'flags' field on the stack in
+ sms_exec() is uninitialised, causing it to use the wrong protocol
+ in some cases. This patch correctly initializes the flags fields.
+ (2) Secondly, when disconnect supervision is not working or
+ inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was
+ failing to terminate the call after it sent the REL(ease) message
+ and the peer stopped talking to it. This patch fixes the code to
+ handle the 'bad stop bit' message more gracefully in that case,
+ and hang up the call. Review:
+ https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close
+ Reported by: David Woodhouse patches: asterisk-fix-sms.patch
+ uploaded by David Woodhouse (License 5754)
+
+2014-04-18 17:12 +0000 [r412585] Rusty Newton <rnewton at digium.com>
+
+ * sounds/sounds.xml, sounds/Makefile: sounds: Fix Sounds Makefile
+ and XML that didn't support new sound prompt sets In
+ sounds/Makefile 1 Adds and moves some lines necessary for the
+ en_GB core set. I'm just following how the other sets are defined
+ here. 2 removes the ES extra sounds related lines as we don't
+ have ES extra sound sets. In sounds/sounds.xml 3 Adds
+ <support_level> definitions to all the sound sets as we have
+ these defined in 11,12,Trunk, but not in 1.8 4 Adds member
+ definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in
+ extra sound sets ASTERISK-23550 Reported by: Rusty Newton Review:
+ https://reviewboard.asterisk.org/r/3464/
+
+2014-04-17 20:23 +0000 [r412480] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_oss.c: channels/chan_oss: Fix compilation problem
+ on SmartOS/Illumos/SunOS THis patch fixes an issue in chan_oss
+ when building on certain platforms. It ensures that soundcard.h
+ is found. Review: https://reviewboard.asterisk.org/r/3426 Note
+ that this patch is a part of the patch on ASTERISK-23576; the
+ Makefile portion only applies to Asterisk 11+. (issue
+ ASTERISK-23576) Reported by: Sebastian Wiedenroth patches:
+ fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
+
+2014-04-15 15:21 +0000 [r412328] Jonathan Rose <jrose at digium.com>
+
+ * configs/sip.conf.sample, channels/chan_sip.c: Reverting r411189
+ so that it can be put up for public review --- r411189 | jrose |
+ 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip:
+ Send real CallerID information with P-Assserted-Identity
+ (RFC-3325) Prior to this patch, the P-Asserted-Identity header
+ would include anonymous caller id information which seems to go
+ against the point of the P-Asserted-Identity header. Now the real
+ caller ID information will be included in this header. Also, no
+ privacy header would be included. This patch adds 'Privacy: id'
+ to outgoing SIP messages that include the P-Asserted-Identity
+ header. (closes issue AST-1301) ---
+
+2014-04-11 21:37 +0000 [r412225] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_stack.c: app_stack: Add missing unlock in off-nominal
+ path of STACK_PEEK function. ASTERISK-23620 #close Reported by:
+ Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch
+ (license #5021) patch uploaded by Bradley Watkins
+
+2014-04-11 01:33 +0000 [r412114] Matthew Jordan <mjordan at digium.com>
+
+ * main/utils.c, main/astobj2.c, contrib/scripts/refcounter.py
+ (added), main/asterisk.c, build_tools/cflags.xml,
+ include/asterisk/utils.h, channels/chan_sip.c,
+ include/asterisk/astobj2.h, main/logger.c: main/astobj2: Make
+ REF_DEBUG a menuselect item; improve REF_DEBUG output This patch
+ does the following: (1) It makes REF_DEBUG a meneselect item.
+ Enabling REF_DEBUG now enables REF_DEBUG globally throughout
+ Asterisk. (2) The ref debug log file is now created in the
+ AST_LOG_DIR directory. Every run will now blow away the previous
+ run (as large ref files sometimes caused issues). We now also no
+ longer open/close the file on each write, instead relying on
+ fflush to make sure data gets written to the file (in case the
+ ao2 call being performed is about to cause a crash) (3) It goes
+ with a comma delineated format for the ref debug file. This makes
+ parsing much easier. This also now includes the thread ID of the
+ thread that caused ref change. (4) A new python script instead
+ for refcounting has been added in the contrib/scripts folder.
+ Review: https://reviewboard.asterisk.org/r/3377/
+
+2014-04-08 21:15 +0000 [r411960-411964] Richard Mudgett <rmudgett at digium.com>
+
+ * main/asterisk.c: Internal timing: Add notice that the -I and
+ internal_timing option are no longer needed. Add notice messages
+ during execution that the -I command line option and the
+ astersik.conf internal_timing option are no longer needed. The
+ internal timing functionality is now always enabled if there is a
+ timing module loaded. NOTE: Since the command line options and
+ the asterisk.conf config file are processed before the logging
+ system is initialized, the messages are output to stderr. Change
+ requested as a result of asterisk-dev list comments about the
+ commit for ASTERISK-22846 that removed the -I and internal_timing
+ options. Review: https://reviewboard.asterisk.org/r/3423/
+
+ * main/config.c: config: Fix CB_ADD_LEN() to work as originally
+ intended. Fix a long standing bug in CB_ADD_LEN() behaving like
+ CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes
+
+2014-04-07 14:45 +0000 [r411807] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * configs/res_odbc.conf.sample: configs: Clean up long line and
+ typo in res_odbc.conf.sample.
+
+2014-04-04 18:32 +0000 [r411715] Richard Mudgett <rmudgett at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, main/channel.c,
+ channels/chan_sip.c, configs/asterisk.conf.sample, UPGRADE.txt:
+ internal_timing: Remove the option and always make it enabled if
+ a timing module is loaded. The masquerade supertest frequently
+ fails because either the local channel chain doesn't completely
+ optimize out or the DTMF handshake doesn't completely get
+ accross. Local channel optimization requires frames flowing to
+ trigger when optimization can happen. When optimization happens
+ the media frame that triggered the optimization is dropped.
+ Sending DTMF requires frames to flow in the other direction for
+ timing purposes while sending nothing. If internal timing is not
+ enabled when MOH is playing, Asterisk switches to received timing
+ when an audio frame is received. With optimization dropping media
+ frames and MOH not sending frames unless it receives frames,
+ occasionaly there are no more frames being passed and the test
+ fails. * The asterisk command line -I option and the
+ asterisk.conf internal_timing option are removed. Asterisk now
+ always uses internal timing when needed if any timing module is
+ loaded. The issue ASTERISK-14861 did this quite awhile ago in
+ v1.4 but effectively is broken if other internal timing modules
+ besides DAHDI are used. The ast_read_generator_actions() now only
+ does received timing if it has no choice for frame generators
+ like MOH, silence, and playback streaming. * Cleaned up some code
+ dealing with frame generators in ast_deactivate_generator(),
+ generator_write_format_change(), ast_activate_generator(), and
+ ast_channel_stop_silence_generator(). ASTERISK-22846 #close
+ Reported by: Matt Jordan Review:
+ https://reviewboard.asterisk.org/r/3414/
+
+2014-04-01 16:48 +0000 [r411584] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_queue.c: app_queue: Fix a bug where realtime members
+ would be deleted during reload causing waiting callers to get
+ ejected. This patch causes realtime queue members to remain in
+ queues during the reload process. Previously these members would
+ be removed causing any waiting callers to be ejected from the
+ queue with a reason of "EXITEMPTY". ASTERISK-23547 #close
+ ASTERISK-23547 #comment Patch
+ app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo
+ Rossi (license 6409) Review:
+ https://reviewboard.asterisk.org/r/3404/
+
+2013-04-23 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.27.0 Released.
+
+2013-04-21 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.27.0-rc2 Released.
+
+ * chan_sip: Add sendrpid trust options
+
+ In r411189, some behavior was changed which made sendrpid behavior
+ act in a more trusting manner by sending full user data for peers
+ set with private caller presence in P-Asserted-Identity headers.
+ Since this changed long time expected behaviors, we decided to pull
+ that patch when that was pointed out by the community. Instead, this
+ patch provides a trust_id_outbound setting which will expose the data
+ per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers
+ at all if set to 'no'. By default trust_id_outbound will be set to
+ 'legacy' which will preserve the behavior prior to these patches.
+ Extra special thanks to Walter Doekes for providing advice and
+ feedback.
+
+2014-03-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 1.8.27.0-rc1 Released.
+
+2014-03-28 16:16 +0000 [r411462] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/http.c, main/tcptls.c, main/manager.c: http: response body
+ often missing after specific request This patch works around a
+ problem with the HTTP body being dropped from the response to a
+ specific client and under specific circumstances: a) Client
+ request comes from node.js user agent "Shred" via use of
+ swagger-client library. b) Asterisk and Client are *not* on the
+ same host or TCP/IP stack In testing this problem, it has been
+ determined that the write of the HTTP body is lost, even if the
+ data is written using low level write function. The only solution
+ found is to instruct the TCP stack with the shutdown function to
+ flush the last write and finish the transmission. See review for
+ more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
+ Reported by: Sam Galarneau Review:
+ https://reviewboard.asterisk.org/r/3402/
+
+2014-03-28 15:42 +0000 [r411372-411457] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between 1.4
+ and 1.8+ systems.
+
+ * res/res_config_odbc.c, res/res_odbc.exports.in, UPGRADE.txt,
+ res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h: res_config_odbc/res_odbc: Fix
+ handling of non-text columns updates with empty values. This
+ patch fixes setting nullable integer columns to NULL instead of
+ an empty string, which fails for PostgreSQL, for example. The
+ current code is supposed to do so, but the check is broken. The
+ patch also allows the first column in the list to be a nullable
+ integer. This patch also adds a compatibility setting in
+ res_odbc.conf, allow_empty_string_in_nontext. It is enabled by
+ default. It should be disabled for database backends (such as
+ PostgreSQL) that require NULL instead of an empty string for
+ Integer columns. Review: https://reviewboard.asterisk.org/r/3375
+ (issue ASTERISK-23459) Reported by: zvision patches:
+ res_config_odbc.diff uploaded by zvision (License 5755)
+
+ * channels/sip/include/sip.h: chan_sip: Add MESSAGE request to
+ allowed methods The allowed methods advertised by chan_sip did
+ not previously note the MESSAGE request. Even in Asterisk 1.8, we
+ do accept in-dialog MESSAGE requests; we should advertise that we
+ support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
+ #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
+ Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
+ Review: https://reviewboard.asterisk.org/r/3396/
+
+2014-03-27 19:06 +0000 [r411313] Corey Farrell <git at cfware.com>
+
+ * funcs/func_groupcount.c, funcs/func_callcompletion.c,
+ funcs/func_pitchshift.c, funcs/func_odbc.c, funcs/func_volume.c,
+ funcs/func_frame_trace.c, funcs/func_channel.c,
+ funcs/func_blacklist.c, funcs/func_callerid.c, apps/app_stack.c,
+ res/res_calendar.c, apps/app_jack.c, funcs/func_speex.c,
+ funcs/func_dialplan.c, channels/chan_sip.c, funcs/func_math.c,
+ apps/app_readexten.c, funcs/func_strings.c, res/res_jabber.c,
+ channels/chan_iax2.c, res/res_mutestream.c, funcs/func_global.c,
+ apps/app_speech_utils.c: Fix dialplan function NULL channel
+ safety issues (closes issue ASTERISK-23391) Reported by: Corey
+ Farrell Review: https://reviewboard.asterisk.org/r/3386/
+
+2014-03-26 22:43 +0000 [r411243] Joshua Colp <jcolp at digium.com>
+
+ * main/say.c: say: Fix a bug where SayNumber in Polish tries to
+ play incorrect sound. This change fixes a bug where calling
+ SayNumber with a number divisible by 100 using the Polish
+ language would cause the code to attempt to play a sound file
+ with an empty name. (closes issue ASTERISK-23509) Reported by:
+ zvision Review: https://reviewboard.asterisk.org/r/3378/
+
+2014-03-26 15:50 +0000 [r411189] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Send real
+ CallerID information with P-Assserted-Identity (RFC-3325) Prior
+ too this patch, the P-Asserted-Identity header would include
+ anonymous caller id information which seems to go against the
+ point of the P-Asserted-Identity header. Now the real caller ID
+ information will be included in this header. Also, no privacy
+ header would be included. This patch adds 'Privacy: id' to
+ outgoing SIP messages that include the P-Asserted-Identity
+ header. (closes issue AST-1301)
+
+2014-03-25 15:50 +0000 [r411088] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
+ update_provisional_keepalive() is called while
+ send_provisional_keepalive_full() is waiting on the PVT lock,
+ then pvt->provisional_keepalive_sched_id will be changed to a new
+ sched_id value by update_provisional_keepalive(), but that new
+ sched_id then may be overwritten with -1 by
+ send_provisional_keepalive_full(), killing the pvt's reference to
+ a schedule and "leaking" the reference. (closes issue
+ ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
+ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
+ Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
+ (license 5012)
+
+2014-03-24 21:36 +0000 [r411021] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Always use fromdomain if set for
+ domain, even if callerid is set to restricted. (closes issue
+ ASTERISK-20841) Reported by: Kelly Goedert
+
+2014-03-17 21:54 +0000 [r410710-410748] Russ Meyerriecks <rmeyerreicks at digium.com>
+
+ * main/callerid.c: !fixup: callerid: Logic error in checksum
+ processing Fixes syntax error in previous commit :-(
+
+ * main/callerid.c: callerid: Logic error in checksum processing
+ Callerid checksum-ing was being handled incorrectly here. When
+ the checksum is calculated to be 0x00, it will perform 0x100-0x00
+ which results in 0x100. This value will then fail the otherwise
+ correct callerid message. This patch changes the logic to simply
+ add the calculated checksum to the transmitted 2's compliment
+ checksum. Review: https://reviewboard.asterisk.org/r/3356/
+ (closes issue ASTERISK-23488)
+
+2014-03-10 17:00 +0000 [r410380] Richard Mudgett <rmudgett at digium.com>
+
+ * main/http.c: AST-2014-001: Stack overflow in HTTP processing of
+ Cookie headers. Sending a HTTP request that is handled by
+ Asterisk with a large number of Cookie headers could overflow the
+ stack. Another vulnerability along similar lines is any HTTP
+ request with a ridiculous number of headers in the request could
+ exhaust system memory. (closes issue ASTERISK-23340) Reported by:
+ Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
+ Manuel Sadosky, Buenos Aires, Argentina
+
+2014-03-10 13:15 +0000 [r410308] Kinsey Moore <kmoore at digium.com>
+
+ * channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
+ session timers request This change allows chan_sip to avoid
+ creation of the channel and consumption of associated file
+ descriptors altogether if the inbound request is going to be
+ rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
+ Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
+ Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
+ Corey Farrell (license 5909)
+
+2014-03-07 22:50 +0000 [r410224] Corey Farrell <git at cfware.com>
+
+ * channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
+ unload_module and do_monitor Release monlock before calling
+ pthread_join. This ensures do_monitor cannot freeze by locking
+ monlock during module unload. (closes issue ASTERISK-21406)
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/3284/
+
+2014-03-07 04:35 +0000 [r410105] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Allow static realtime members to
+ be qualified during module load. When a static realtime peer with
+ qualify=yes is loaded, Asterisk will fail to send an OPTIONS
+ request due to the lastms being equal to 0. This results in the
+ peer being unable to receive calls from Asterisk because the
+ status is permanently UNKNOWN. This patch allows an OPTIONS
+ request to be sent during module load by ignoring the lastms
+ value on startup only. Review:
+ https://reviewboard.asterisk.org/r/3294/ (closes issue
+ ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
+ wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
+ Peirce (license 6112)
+
+2014-03-06 23:01 +0000 [r410043] Russell Bryant <russell at russellbryant.com>
+
+ * res/res_musiconhold.c: moh: fix a refcount error with realtime
+ MOH I observed a crash in res_musiconhold on an Asterisk 11
+ system using realtime MOH. Investigation of the backtrace showed
+ a corrupt mohclass, implying that it got destroyed before the
+ code expected it to. I went looking for reference counting errors
+ that could have caused this crash and this patch this result. It
+ contains 2 changes. 1) Remove a usless block of code that was
+ impossible to reach. There was even a comment indicating that it
+ was impossible to reach. The conditional includes
+ "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
+ inside of an if block with the opposite check
+ "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
+ good reason to keep it around. 2) A similar block to #1 contained
+ a reference counting error. It stores state->class in the local
+ variable mohclass without increasing its reference count. The
+ reference count on mohclass is decremented at the end of the
+ function. This block of code probably very rarely runs, which
+ would help explain why this system was working fine for many
+ months before experiencing a crash. Review:
+ https://reviewboard.asterisk.org/r/3282/
+
+2014-03-05 20:31 +0000 [r409916] Kinsey Moore <kmoore at digium.com>
+
+ * main/config.c: config: Fix inverted test The test of the result
+ of the stat() call was inverted such that its output was only
+ used if the call failed. This inverts the test so that the output
+ of stat() is used correctly. This was causing full reloads on
+ unchanged files. (closes issue ASTERISK-23383) Reported by: David
+ Woolley
+
+2014-03-05 16:50 +0000 [r409833] David M. Lee <dlee at digium.com>
+
+ * main/config.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: Corrected cross-platform stat nanosecond code When
+ nanosecond time resolution was added for identifying config file
+ changes, it didn't cover all of the myriad of ways that one might
+ obtain nanosecond time resolution off of struct stat. Rather than
+ complicate the #if even further figuring out one system from the
+ next, this patch directly tests for the three struct members I
+ know about today, and #ifdef's accordingly. Review:
+ https://reviewboard.asterisk.org/r/3273/
+
+2014-03-05 12:04 +0000 [r409777] Sean Bright <sean at malleable.com>
+
+ * contrib/scripts/astgenkey, contrib/scripts/astgenkey.8: Fix
+ references to 'keys' CLI commands in astgenkey
+
+2014-03-05 05:10 +0000 [r409705] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Add update_peer function to
+ unistim_rtp_glue, improve other unistim_rtp_glue functions
+ conforming to other channel drivers. Do not forget auto-detected
+ and user-selected phone settings on 'unistim reload'
+
+2014-03-04 19:32 +0000 [r409623] Michael L. Young <elgueromexicano at gmail.com>
+
+ * funcs/func_audiohookinherit.c: func_audiohookinheritance: Check
+ If A Channel Was Specified This patch prevents a crash when using
+ the function audiohookinheritance without setting the channel.
+ (closes issue ASTERISK-23104) Reported by: Joel Vandal Tested by:
+ Joel Vandal Patches:
+ asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
+ Michael L. Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/3272/
+
+2014-03-04 16:50 +0000 [r409521-409566] Kinsey Moore <kmoore at digium.com>
+
+ * main/astobj2.c: AO2: Add an assert for bad objects This adds an
+ assert that will only be active if Asterisk is compiled with
+ DO_CRASH and allows the testsuite to fail tests that would
+ otherwise require log file parsing.
+
+ * main/rtp_engine.c: rtp_engine: Clean up after a failed remote
+ bridge Upon failure of an INVITE transaction meant to initiate a
+ remote native bridge, rtp_engine.c would not clean up
+ non-reference-counted bridge instance pointers leaving a dangling
+ pointer which was being used to perform a local native bridge
+ after the other channel had hung up. This lead to dereferencing
+ into freed memory and plenty of AO2 errors. This change allows
+ the remote native bridge loop to clean up properly when the
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