[svn-commits] bebuild: tag 11.10.0-rc1 r414419 - /tags/11.10.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu May 22 11:19:38 CDT 2014
Author: bebuild
Date: Thu May 22 11:19:32 2014
New Revision: 414419
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=414419
Log:
Importing files for 11.10.0-rc1 release.
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tags/11.10.0-rc1/.version (with props)
tags/11.10.0-rc1/ChangeLog (with props)
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--- tags/11.10.0-rc1/ChangeLog (added)
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+2014-05-22 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.10.0-rc1 Released.
+
+2014-05-22 15:50 +0000 [r414402] 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/ ........ Merged
+ revisions 414401 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-22 13:59 +0000 [r414346] Matthew Jordan <mjordan at digium.com>
+
+ * UPGRADE.txt, /: UPGRADE: Add note for REF_DEBUG flag ........
+ Merged revisions 414345 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-21 22:05 +0000 [r414270] 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. ........ Merged revisions 414269 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-21 19:05 +0000 [r414215] 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 ASTEIRSK-23650 #close Review:
+ https://reviewboard.asterisk.org/r/3539/ ........ Merged
+ revisions 414214 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-19 13:37 +0000 [r414153] 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 ........ Merged revisions
+ 414152 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-16 20:03 +0000 [r413992-414068] 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/ ........ Merged
+ revisions 414067 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/sig_pri.c: sig_pri.c: Pull the pri_dchannel()
+ PRI_EVENT_RING case into its own function. * Populate the
+ CALLERID(ani2) value (and the special CALLINGANI2 channel
+ variable) with the ANI2 value in addition to the PRI specific
+ ANI2 channel variable.
+
+ * /, 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. ........ Merged
+ revisions 413991 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-14 15:31 +0000 [r413895] 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/ ........ Merged
+ revisions 413894 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-13 17:40 +0000 [r413876] Jonathan Rose <jrose at digium.com>
+
+ * channels/chan_sip.c: chan_sip: Add TLS and SRTP status to CLI
+ command 'sip show channel' ASTERISK-23564 #close Reported by:
+ Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/
+
+2014-05-13 14:34 +0000 [r413788-413838] 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/ ........ Merged
+ revisions 413832 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_format_attr_h264.c: h264: Fix H264 SDP payload format.
+ https://tools.ietf.org/html/rfc3984#section-8.1 says
+ profile-level-id takes 3 bytes in base16 (6 hex digits). This
+ fixes video setup in certain cases. ASTERISK-23664 #close
+ ASTERISK-23664 #comment Patch r3530.patch uploaded by Guillaume
+ Maudoux. Review: https://reviewboard.asterisk.org/r/3530/
+
+ * 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/ ........ Merged
+ revisions 413787 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-12 23:48 +0000 [r413765] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/sig_pri.c, /, configure,
+ include/asterisk/autoconfig.h.in, configure.ac:
+ 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/
+ ........ Merged revisions 413714 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-12 22:02 +0000 [r413710] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_chanspy.c: app_chanspy: Fix a test that was failing on
+ account of r413551 ASTERISK-23381 #close ASTERISK-23381 #comment
+ Reported by: Robert Moss Review:
+ https://reviewboard.asterisk.org/r/3505/
+
+2014-05-09 23:08 +0000 [r413587-413595] Kinsey Moore <kmoore at digium.com>
+
+ * /, funcs/func_env.c: Fix 32bit build for func_env ........ Merged
+ revisions 413592 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * res/res_fax_spandsp.c, res/res_odbc.c, main/asterisk.c,
+ res/res_calendar.c, apps/app_voicemail.c,
+ channels/chan_unistim.c, main/ccss.c, funcs/func_sysinfo.c,
+ main/utils.c, res/res_musiconhold.c, channels/chan_gtalk.c,
+ res/res_jabber.c, res/res_http_websocket.c,
+ res/res_format_attr_h264.c, main/enum.c, main/io.c,
+ main/channel.c, res/ael/pval.c, channels/chan_phone.c,
+ main/manager.c, res/res_config_odbc.c, apps/app_minivm.c,
+ channels/chan_motif.c, res/res_agi.c, main/logger.c, main/app.c,
+ apps/app_confbridge.c, res/res_format_attr_silk.c,
+ channels/chan_mgcp.c, apps/app_adsiprog.c,
+ res/res_rtp_asterisk.c, main/stun.c,
+ res/res_calendar_icalendar.c, channels/chan_sip.c,
+ apps/app_festival.c, res/res_fax.c, main/translate.c,
+ main/slinfactory.c, res/res_crypto.c, main/acl.c,
+ apps/app_queue.c, apps/app_getcpeid.c, channels/sig_pri.c,
+ res/res_srtp.c, channels/chan_jingle.c, res/res_corosync.c,
+ res/res_stun_monitor.c, main/abstract_jb.c, main/callerid.c,
+ main/file.c, main/config_options.c, main/adsi.c, main/event.c,
+ pbx/pbx_dundi.c, apps/app_sms.c, channels/sip/config_parser.c,
+ apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/xmldoc.c,
+ main/udptl.c, main/format.c, cel/cel_pgsql.c, main/frame.c,
+ channels/chan_local.c, main/rtp_engine.c, main/security_events.c,
+ /, funcs/func_env.c, res/res_timing_dahdi.c,
+ bridges/bridge_softmix.c, main/devicestate.c,
+ cdr/cdr_adaptive_odbc.c, res/res_calendar_caldav.c,
+ main/taskprocessor.c, res/res_xmpp.c, res/res_format_attr_h263.c,
+ channels/chan_iax2.c, res/res_pktccops.c, main/config.c,
+ main/loader.c, main/cli.c, res/res_format_attr_celt.c,
+ apps/confbridge/conf_config_parser.c, funcs/func_hangupcause.c,
+ channels/chan_dahdi.c, cel/cel_odbc.c, channels/sig_analog.c,
+ include/asterisk/astobj.h, channels/chan_skinny.c,
+ formats/format_pcm.c, main/features.c, apps/app_dumpchan.c,
+ funcs/func_srv.c, channels/chan_alsa.c, main/stdtime/localtime.c,
+ main/bridging.c, channels/sip/include/sip.h, main/sched.c,
+ apps/app_dial.c, res/res_calendar_exchange.c, main/pbx.c,
+ pbx/dundi-parser.c, main/aoc.c, main/cel.c,
+ channels/iax2-parser.c, res/res_calendar_ews.c,
+ res/res_monitor.c, main/netsock.c, main/data.c, main/audiohook.c,
+ funcs/func_iconv.c, pbx/pbx_config.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. ........ Merged revisions 413586 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-09 16:10 +0000 [r413551] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_chanspy.c: app_chanspy: Fix a bug where Barge mode could
+ fail If the barge audiohook was attached prior to the spyee and
+ its peer actually being bridged, the audiohook would not be
+ applied and the connected peer would not be able to hear audio
+ from the spy when the spy is in barge mode. (closes issue
+ ASTERISK-23381) Reported by: Robert Moss Review:
+ https://reviewboard.asterisk.org/r/3505/
+
+2014-05-08 00:34 +0000 [r413486] Joshua Colp <jcolp at digium.com>
+
+ * main/manager.c, /, apps/app_queue.c: app_queue: Extend
+ documentation for various Manager actions and events. ........
+ Merged revisions 413485 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-07 20:29 +0000 [r413451] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/app_confbridge.c: app_confbridge: Fix ref leak in CLI
+ "confbridge kick" command. Fixed ref leak in the CLI "confbridge
+ kick" command when the channel to be kicked was not in the
+ conference.
+
+2014-05-07 17:48 +0000 [r413397] 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) ........ Merged revisions 413396 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-06 17:01 +0000 [r413305] 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) ........ Merged revisions 413304 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-05-02 20:25 +0000 [r413225-413251] 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) ........ Merged revisions 413241 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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 ........ Merged
+ revisions 413224 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-30 20:26 +0000 [r413139] Richard Mudgett <rmudgett at digium.com>
+
+ * main/message.c, channels/chan_sip.c, include/asterisk/message.h:
+ chan_sip.c: Fixed off-nominal message iterator ref count and
+ alloc fail issues. * Fixed early exit in sip_msg_send() not
+ destroying the message iterator. * Made
+ ast_msg_var_iterator_next() and ast_msg_var_iterator_destroy()
+ tolerant of a NULL iter parameter in case
+ ast_msg_var_iterator_init() fails. * Made
+ ast_msg_var_iterator_destroy() clean up any current message data
+ ref. * Made struct ast_msg_var_iterator,
+ ast_msg_var_iterator_init(), ast_msg_var_iterator_next(),
+ ast_msg_var_unref_current(), and ast_msg_var_iterator_destroy()
+ use iter instead of i.
+
+2014-04-30 13:04 +0000 [r413123] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_http_websocket.c: Websocket: Add session locking and
+ delay close This resolves a race condition where data could be
+ written to a NULL FILE pointer causing a crash as a websocket
+ connection was in the process of shutting down by adding locking
+ to websocket session writes and by deferring session teardown
+ until session destruction. (closes issue ASTERISK-23605) Review:
+ https://reviewboard.asterisk.org/r/3481/ Reported by: Matt Jordan
+
+2014-04-25 17:47 +0000 [r413008] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Add support for DTLS
+ handshake retransmissions On congested networks, it is possible
+ for the DTLS handshake messages to get lost. This patch adds a
+ timer to res_rtp_asterisk that will periodically check to see if
+ the handshake has succeeded. If not, it will retransmit the DTLS
+ handshake. Review: https://reviewboard.asterisk.org/r/3337
+ ASTERISK-23649 #close Reported by: Nitesh Bansal patches:
+ dtls_retransmission.patch uploaded by Nitesh Bansal (License
+ 6418)
+
+2014-04-23 17:51 +0000 [r412923] 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. ........ Merged revisions 412922 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-21 17:53 +0000 [r412767-412822] 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 ........ Merged revisions
+ 412821 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, CHANGES: Typo in CHANGES ........ Merged revisions 412764 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-21 16:13 +0000 [r412748] 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/ ........ Merged
+ revisions 412745 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-21 15:51 +0000 [r412746] Jonathan Rose <jrose at digium.com>
+
+ * configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h, /,
+ channels/chan_sip.c: 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/ ........ Merged
+ revisions 412744 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-21 08:29 +0000 [r412712] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c: Fix wrong dialtone. The "modulation"
+ should not be referenced for tone+tone as it refers to the on-off
+ characteristic - this often resulted in a single tone rather than
+ the multitone as in the UK.
+
+2014-04-19 01:02 +0000 [r412656] 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) ........ Merged
+ revisions 412655 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-18 17:15 +0000 [r412586] 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 member
+ definitons for EN_AU, EN_GB, IT for core sound sets, and EN_GB in
+ extra sound sets ASTERISK-23550 #close Review:
+ https://reviewboard.asterisk.org/r/3464/
+
+2014-04-17 20:06 +0000 [r412468] Matthew Jordan <mjordan at digium.com>
+
+ * channels/chan_oss.c, main/Makefile: main/Makefile: Fix build
+ failure on SmartOS/Illumos/SunOS This patch fixes two issues when
+ building on SmartOS: - channels/chan_oss.c: it makes sure
+ soundcard.h is found - main/Makefile: only use
+ "-Wl,--version-script" when GNU LD is used as the Sun Linker
+ doesn't support that. Similar checks are already used elswhere in
+ the Makefile Review: https://reviewboard.asterisk.org/r/3426
+ ASTERISK-23576 #close Reported by: Sebastian Wiedenroth patches:
+ fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597)
+
+2014-04-15 16:23 +0000 [r412348] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/chan_sip.c: chan_sip.c: Moved some sip_pvt unrefs after
+ their last use. * Moved sip_pvt unref in ast_hangup() and
+ handle_request_do() to the end of the function. The unref needs
+ to happen after the last use of the pointer.
+
+2014-04-15 15:40 +0000 [r412329] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample: 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) --- ........ Merged revisions 412328 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-14 15:50 +0000 [r412305] Corey Farrell <git at cfware.com>
+
+ * main/autoservice.c: autoservice: fix reference leak of logger
+ callid. autoservice acquires a local reference to the logger
+ callid of each channel in a loop. This local reference was not
+ released, causing the callid of every channel in autoservice to
+ leak. This change moves the callid unref inside the loop.
+ ASTERISK-23616 #close Reported by: ibercom
+
+2014-04-11 21:38 +0000 [r412226] 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 ........ Merged
+ revisions 412225 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-11 02:10 +0000 [r412115] Matthew Jordan <mjordan at digium.com>
+
+ * /, channels/chan_sip.c, channels/sip/security_events.c,
+ include/asterisk/astobj2.h, main/utils.c, main/astobj2.c,
+ contrib/scripts/refcounter.py (added), main/asterisk.c,
+ build_tools/cflags.xml: 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/ ........ Merged
+ revisions 412114 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-08 21:20 +0000 [r411944-411974] 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/
+ ........ Merged revisions 411964 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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
+ ........ Merged revisions 411960 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/confbridge/conf_config_parser.c: app_confbridge: Fix
+ confbridge.conf dsp_talking_threshold option setting wrong
+ parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported
+ by: John Knott
+
+2014-04-07 14:48 +0000 [r411808] Walter Doekes <walter+asterisk at wjd.nu>
+
+ * configs/res_odbc.conf.sample, /, UPGRADE.txt: configs: Clean up
+ long line and typo in res_odbc.conf.sample. ........ Merged
+ revisions 411807 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-04 18:46 +0000 [r411716] 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/ ........ Merged
+ revisions 411715 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-01 20:43 +0000 [r411633] Corey Farrell <git at cfware.com>
+
+ * utils/extconf.c, include/asterisk/options.h, main/asterisk.c,
+ apps/app_voicemail.c, main/channel.c: app_voicemail: fix missing
+ symbol ASTERISK-23391 caused a regression where the symbol
+ 'defaultlanguage' was used by app_voicemail but not exported by
+ main/asterisk. This change renames the variable to
+ ast_defaultlanguage. The variable was already renamed in Asterisk
+ 12+. (closes issue ASTERISK-23559) Reported by: Corey Farrell
+ Review: https://reviewboard.asterisk.org/r/3408/
+
+2014-04-01 16:49 +0000 [r411585] 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/ ........ Merged
+ revisions 411584 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-04-23 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.9.0 Released.
+
+2014-04-21 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.9.0-rc3 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.
+
+2014-04-14 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.9.0-rc2 Released.
+
+ * autoservice: fix reference leak of logger callid.
+
+ autoservice acquires a local reference to the logger callid of each
+ channel in a loop. This local reference was not released, causing
+ the callid of every channel in autoservice to leak. This change moves
+ the callid unref inside the loop.
+
+ ASTERISK-23616 #close
+ Reported by: ibercom
+
+ * app_voicemail: fix missing symbol
+
+ ASTERISK-23391 caused a regression where the symbol 'defaultlanguage'
+ was used by app_voicemail but not exported by main/asterisk. This
+ change renames the variable to ast_defaultlanguage. The variable was
+ already renamed in Asterisk 12+.
+
+ (closes issue ASTERISK-23559)
+ Reported by: Corey Farrell
+
+2014-03-28 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 11.9.0-rc1 Released.
+
+2014-03-28 17:44 +0000 [r411531] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/oochannels.c,
+ addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c,
+ addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+ addons/chan_ooh323.c: process stack command even if gatekeeper
+ client isn't register don't destroy gatekeeper client if it is
+ not started don't destroy gatekeeper client in some sort of
+ gatekeeper errors signal rtp create condition when call cleared
+ before rtp structure created (closes issue ASTERISK-23460)
+ Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch
+ Tested by: Dmitry Melekhov
+
+2014-03-28 16:16 +0000 [r411463] Scott Griepentrog <sgriepentrog at digium.com>
+
+ * main/manager.c, /, main/http.c, main/tcptls.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/ ........ Merged
+ revisions 411462 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-28 15:43 +0000 [r411373-411458] Matthew Jordan <mjordan at digium.com>
+
+ * /, UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between
+ 1.4 and 1.8+ systems. ........ Merged revisions 411457 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/res_odbc.conf.sample, include/asterisk/res_odbc.h,
+ res/res_config_odbc.c, /, res/res_odbc.exports.in, UPGRADE.txt,
+ res/res_odbc.c: 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) ........
+ Merged revisions 411399 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * 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/ ........ Merged
+ revisions 411372 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
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