[svn-commits] bebuild: tag 11.9.0-rc1 r411553 - /tags/11.9.0-rc1/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 28 13:29:55 CDT 2014
Author: bebuild
Date: Fri Mar 28 13:29:47 2014
New Revision: 411553
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411553
Log:
Importing files for 11.9.0-rc1 release.
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tags/11.9.0-rc1/.version (with props)
tags/11.9.0-rc1/ChangeLog (with props)
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+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
+
+2014-03-27 19:13 +0000 [r411310-411314] Corey Farrell <git at cfware.com>
+
+ * funcs/func_strings.c, funcs/func_math.c,
+ funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c,
+ res/res_jabber.c, res/res_mutestream.c, funcs/func_global.c,
+ apps/app_speech_utils.c, apps/confbridge/conf_config_parser.c,
+ funcs/func_pitchshift.c, funcs/func_callcompletion.c,
+ funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c,
+ funcs/func_blacklist.c, funcs/func_channel.c,
+ funcs/func_frame_trace.c, main/features.c, funcs/func_callerid.c,
+ apps/app_stack.c, main/message.c, res/res_calendar.c,
+ apps/app_jack.c, apps/app_voicemail.c, funcs/func_speex.c,
+ funcs/func_dialplan.c, channels/chan_sip.c, /: Fix dialplan
+ function NULL channel safety issues (closes issue ASTERISK-23391)
+ Reported by: Corey Farrell Review:
+ https://reviewboard.asterisk.org/r/3386/ ........ Merged
+ revisions 411313 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/format.c, include/asterisk.h, main/asterisk.c: main/formats:
+ Fix crash in ast_format_cmp during non-clean shutdown. * Backport
+ ast_register_cleanup from Asterisk 12. * Use ast_register_cleanup
+ for format_attr_shutdown. ast_register_cleanup was originally
+ commited in r390122 by dlee. (closes issue ASTERISK-23103)
+ Reported by: JoshE
+
+2014-03-26 22:44 +0000 [r411244] 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/ ........
+ Merged revisions 411243 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-26 15:57 +0000 [r411190] Jonathan Rose <jrose at digium.com>
+
+ * configs/sip.conf.sample, /, channels/chan_sip.c: 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) ........ Merged revisions 411189
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-25 15:52 +0000 [r411089] 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) ........ Merged revisions 411088 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-24 21:37 +0000 [r411022] 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 ........ Merged
+ revisions 411021 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-20 22:46 +0000 [r410965] Jonathan Rose <jrose at digium.com>
+
+ * apps/app_confbridge.c: app_confbridge: Fix bug - users with
+ startmuted set don't start muted (closes issue ASTERISK-23461)
+ Reported by: Chico Manobela Review:
+ https://reviewboard.asterisk.org/r/3373/
+
+2014-03-18 11:50 +0000 [r410829] Sean Bright <sean at malleable.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: Use g711_free() when
+ available. Per Johann Steinwendtner on the asterisk-dev mailing
+ list:
+ http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
+ g711_free() was introduced in spandsp 0.0.6pre4 and
+ g711_release() became a noop. I opted not to remove the call to
+ g711_release() since it is harmless and to call g711_free() if we
+ have a sufficiently recent version of spandsp. (issue
+ ASTERISK-20149) Reported by: Alexandr Gordeev
+
+2014-03-17 21:55 +0000 [r410717-410749] Russ Meyerriecks <rmeyerreicks at digium.com>
+
+ * /, main/callerid.c: !fixup: callerid: Logic error in checksum
+ processing Fixes syntax error in previous commit :-( ........
+ Merged revisions 410748 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, 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) ........ Merged revisions 410710
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-14 21:12 +0000 [r410609] Jonathan Rose <jrose at digium.com>
+
+ * main/manager.c: manager: fix memory leak in manager_add_filter
+ function (closes issue ASTERISK-23420) Reported by: Etienne
+ Lessard Patches: manager_eventfilter_leak uploaded by Etienne
+ Lessard (license 6394)
+
+2014-03-14 20:53 +0000 [r410556-410606] Mark Michelson <mmichelson at digium.com>
+
+ * main/db.c: Remove an extra ast_cond_wait() that slipped through
+ the patch.
+
+ * main/db.c: Prevent delayed astdb syncs. The syncing thread sleeps
+ for a second before waiting to be told to attempt to sync again.
+ If a signal were sent during this sleeping period, we would end
+ up having to wait until the next sync signal occurred in order to
+ sync up the astdb. This code rearrangement also ensures that any
+ pending transactions will be synced prior to Asterisk shutting
+ down. Patches: db_sync.patch by John Hardin (License #6512)
+
+2014-03-12 18:35 +0000 [r410490] Richard Mudgett <rmudgett at digium.com>
+
+ * apps/confbridge/conf_state.c,
+ apps/confbridge/conf_state_single.c,
+ apps/confbridge/conf_state_inactive.c,
+ apps/confbridge/conf_state_single_marked.c: app_confbridge: Make
+ explicitly stop MOH if a user is kicked or hangs up while MOH is
+ playing. When MOH is playing to a user in a conference and the
+ user is kicked or hangs up from the conference then the AMI
+ MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
+ MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
+ by: Benjamin Keith Ford Review:
+ https://reviewboard.asterisk.org/r/3306/
+
+2014-03-10 17:09 +0000 [r410381] 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 ........ Merged revisions
+ 410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-10 13:18 +0000 [r410311] 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) ........ Merged revisions 410308
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-07 22:52 +0000 [r410225] 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/ ........ Merged
+ revisions 410224 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-07 04:38 +0000 [r410106] 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) ........ Merged revisions 410105 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-06 23:15 +0000 [r410044] 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/ ........ Merged
+ revisions 410043 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-06 01:58 +0000 [r409990] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_fax_spandsp.c: res_fax_spandsp: Fix crash when passing
+ ulaw/alaw data to spandsp When acting as a T.38 fax gateway,
+ res_fax_spandsp would at times cause a crash in libspandsp. This
+ would occur when, during fax tone detection, a ulaw/alaw frame
+ would be passed to modem_connect_tones_rx. That particular
+ routine expects the data to be in slin format. This patch looks
+ at the frame type and, if the data is ulaw/alaw, converts the
+ format to slin before passing it to modem_connect_tones_rx.
+ Review: https://reviewboard.asterisk.org/r/3296 (closes issue
+ ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal
+ Rybarik patches: spandsp_g711decode.diff uploaded by Michal
+ Rybarik (license 6578)
+
+2014-03-05 20:37 +0000 [r409917] 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 ........ Merged revisions 409916 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 18:45 +0000 [r409886] Mark Michelson <mmichelson at digium.com>
+
+ * funcs/func_presencestate.c: Fix documentation for PRESENCE_STATE
+ to properly illustrate how to create a presence hint. There was a
+ missing comma. This was discovered by Dan Kaplan.
+
+2014-03-05 16:55 +0000 [r409834] 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/ ........ Merged
+ revisions 409833 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 12:04 +0000 [r409778] Sean Bright <sean at malleable.com>
+
+ * /, contrib/scripts/astgenkey, contrib/scripts/astgenkey.8: Fix
+ references to 'keys' CLI commands in astgenkey ........ Merged
+ revisions 409777 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 06:28 +0000 [r409745-409761] Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+ * channels/chan_unistim.c, /: Correct RTP handling in chan_unistim
+ and fix transfer process broken in previous fix: - Fixed too
+ early RTP setup with phone, that cause no ringback tone on caller
+ side - Handle call transfer cancel only in STATE_CALL case
+ (related to ASTERISK-23073) (Reported by: Németh Tamás, niurkin
+ sil)
+
+ * 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' ........
+ Merged revisions 409705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 04:55 +0000 [r409681-409703] Moises Silva <moises.silva at gmail.com>
+
+ * res/res_http_websocket.c: Fix res/res_http_websocket.c build
+ failure in 32bit due to incorrect print format for uint64_t
+
+ * res/res_http_websocket.c: Fix WebRTC over WSS not working Several
+ fixes for the WebSockets implementation in
+ res/res_http_websocket.c * Flush the websocket session FILE* as
+ fwrite() may not actually guarantee sending the data to the
+ network. If we do not flush, it seems that buffering on the SSL
+ socket for outbound messages causes issues * Refactored
+ ast_websocket_read to take into account that SSL file descriptors
+ may be ready to read via fread() but poll() will not actually say
+ so because the data was already read from the network buffers and
+ is now in the libc buffers (closes issue ASTERISK-23099) (closes
+ issue ASTERISK-21930) Review:
+ https://reviewboard.asterisk.org/r/3248/
+
+2014-03-04 19:33 +0000 [r409625] 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/ ........ Merged
+ revisions 409623 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 16:51 +0000 [r409567] 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. ........ Merged revisions
+ 409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 16:40 +0000 [r409565] Jonathan Rose <jrose at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio
+ problems with hold/unhold when using ICE ICE sessions will now be
+ restarted if sessions are changed to use new sets of remote
+ candidates. (closes issue ASTERISK-22911) Reported by: Vytis
+ ValentinaviÄius Review: https://reviewboard.asterisk.org/r/3275/
+
+2014-03-04 15:35 +0000 [r409524] Kinsey Moore <kmoore at digium.com>
+
+ * 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
+ bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy
+ Laine ........ Merged revisions 409521 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 14:52 +0000 [r409473] Sean Bright <sean at malleable.com>
+
+ * /, channels/chan_sip.c: Minor whitespace change to 'sip show
+ peers' output. (closes issue ASTERISK-23406) Reported by: ibercom
+ Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom
+ ........ Merged revisions 409472 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-03 02:07 +0000 [r409362] Matthew Jordan <mjordan at digium.com>
+
+ * /, main/asterisk.c: doxygen: Tweak the link back to ye olde
+ Digium website ........ Merged revisions 409361 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-02 12:26 +0000 [r409344] Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+ * /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a
+ legal option of gcc. Unofficially gcc considers it to be
+ equivalent of -O3. clang chalks on it, though. This commit sets
+ the default optimization flag to be -O3, like gcc actually
+ considered it. Review: https://reviewboard.asterisk.org/r/3280/
+ ........ Merged revisions 409308 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-28 21:30 +0000 [r409255] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Add precautionary p->owner
+ checks. * Add precautionary p->owner checks in sip_hangup(),
+ get_refer_info(), get_also_info(), and
+ interpret_t38_parameters(). * Simplify some tangled logic in
+ get_refer_info(), get_also_info(), and add_rpid(). * Removed some
+ dead code in handle_request_invite(). (closes issue
+ ASTERISK-23323) Reported by: Walter Doekes Patches:
+ issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-11.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-12.x.patch (license #5674)
+ uploaded by wdoekes (modified)
+ issueA23323-more_p_owner_checks-trunk.patch (license #5674)
+ uploaded by wdoekes (modified) ........ Merged revisions 409207
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-28 21:13 +0000 [r409208] Kinsey Moore <kmoore at digium.com>
+
+ * apps/app_queue.c: app_queue: Fix documentation generation The
+ documentation for QueueMemberPaused was causing documentation
+ generation to fail because the documentation for that AMI event
+ was in the wrong location. This moves that documentation the
+ correct location and adds a missing parameter. (closes issue
+ SWDAT-261)
+
+2014-02-28 18:00 +0000 [r409157] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Fix crash in
+ ast_channel_hangupcause_set(). * Fix crash in
+ ast_channel_hangupcause_set() because p->owner not checked before
+ calling. Regression introduced by the fix for ASTERISK-22621.
+ (closes issue ASTERISK-23135) Reported by: OK (issue
+ ASTERISK-23323) Reported by: Walter Doekes ........ Merged
+ revisions 409156 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 19:38 +0000 [r409129-409130] Jonathan Rose <jrose at digium.com>
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: correct build error
+ from r409129 Accidentally placed a declaration below functional
+ code (issue ASTERISK-23213) Reported by: Andrea Suisani Review:
+ https://reviewboard.asterisk.org/r/3256/
+
+ * res/res_rtp_asterisk.c: res_rtp_asterisk: Fix checklist creating
+ problems in ICE sessions Prior to this patch, local candidate
+ lists including SRFLX would fail to start properly when building
+ ICE candidate check lists. This patch fixes that problem by
+ making sure that each SRFLX candidate is associated with the
+ proper base address so that the check list can create matches
+ properly. This patch was written by jcolp. The issue will be left
+ open to await testing by the issue participants. (issue
+ ASTERISK-23213) Reported by: Andrea Suisani Review:
+ https://reviewboard.asterisk.org/r/3256/
+
+2014-02-27 16:24 +0000 [r409083] David M. Lee <dlee at digium.com>
+
+ * /, utils/astman.c: Fix memory stomping bug in astman. This memset
+ complained in dev mod on my Ubuntu box. The memset is both
+ unnecessary and dangerous. At this point, m hasn't been
+ initialized yet, so the memset will write off to whatever address
+ happens to be on the stack at the time. ........ Merged revisions
+ 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 16:03 +0000 [r409053] Corey Farrell <git at cfware.com>
+
+ * /, res/res_fax.c, configs/res_fax.conf.sample: res_fax: Warn that
+ minrate=2400 is not valid for V.27 instead of failing load.
+ Change minrate from 2400 to 4800 on config reload in response to
+ changes from ASTERISK-22790 only. Any config with minrate of 2400
+ that would fail before r405693 will still fail. Comment out many
+ settings in res_fax.conf.sample. The defaults are set in
+ res_fax.c, so setting the same value in sample config does
+ nothing but make the sample config more fragile. (closes issue
+ ASTERISK-23231) Reported by: David Brillert Review:
+ https://reviewboard.asterisk.org/r/3261/ ........ Merged
+ revisions 409052 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 12:47 +0000 [r409002] Matthew Jordan <mjordan at digium.com>
+
+ * main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine:
+ fix crash during remote native bridging when calling get_codecs
+ When two RTP channels are in a remote bridge, the remote bridging
+ loop in rtp_engine will periodically check to see if the two
+ channels can still be bridged. One of the many things it checks
+ is whether or not the codecs have changed on the channel. If the
+ codec has changed, it will break out of the loop to re-determine
+ which type of bridge is appropriate. In order to perform this
+ check, the ast_rtp_glue virtual table's get_codec callback is
+ called for each channel. The callback implementations assume that
+ the channel tech private is valid when called; as such, there has
+ always been some code in place to check whether or not the
+ channel pvt is NULL before calling. However, this check is
+ insufficient. The channels are unlocked during the remote
+ bridging loop. It is possible for a channel to get masqueraded
+ between the check for the pvt being NULL and the actual call to
+ get_codec. When this occurs, the callback is called with a ZOMBIE
+ channel, which now has a NULL pvt. Crash. While this has always
+ been possible in Asterisk 1.8, it is much more likely to occur in
+ Asterisk 11 and later versions due to the timing changes that
+ occur when getting the codec from a channel. Note that this is
+ much more likely to be reproduced on slow, boggy hardware running
+ Asterisk 11 - but fairly rarely otherwise. Also Note: This crash
+ was also caught by the various SIP blind transfer tests, in
+ addition to the bug report Alec filed. Review:
+ https://reviewboard.asterisk.org/r/3247/ (closes issue
+ ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis
+ ........ Merged revisions 409001 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-25 17:43 +0000 [r408877] Rusty Newton <rnewton at digium.com>
+
+ * /, configs/voicemail.conf.sample: configs/voicemail.conf.sample -
+ Make mailcmd sample text more explicit Made the wording a bit
+ more explicit. Didn't really change the meaning. ........ Merged
+ revisions 408876 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-22 17:42 +0000 [r408838] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/chan_ooh323.c: ignore AST_CONTROL_PVT_CAUSE_CODE without
+ any messages (closes issue ASTERISK-23336) Reported by: Alexander
+ Semych
+
+2014-02-22 02:28 +0000 [r408786] Corey Farrell <git at cfware.com>
+
+ * res/ael/pval.c, main/pbx.c, /, utils/extconf.c, utils/conf2ael.c:
+ Remove extra defines of AST_PBX_MAX_STACK. * Ensure
+ AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
+ incorrect function parameters in utils/extconf.c. (closes issue
+ ASTERISK-23141) Reported by: Maxim Review:
+ https://reviewboard.asterisk.org/r/3241/ ........ Merged
+ revisions 408785 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 20:21 +0000 [r408643-408748] Kevin Harwell <kharwell at digium.com>
+
+ * /, apps/app_forkcdr.c: app_forkcdr: ForkCDR v option does not
+ keep CDR variables for subsequent records When the 'v' option is
+ specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is
+ set only for the first CDR in the chain. So ForkCDR works fine
+ with this option only once. After the second and further calls to
+ ForkCDR, CDR variables get cleared on all CDRs besides the first
+ one and moved to the newly forked CDR. It always sets the
+ KEEP_VARS flag on the first CDR in the chain, instead of the most
+ recent CDR which is used as a base to fork a new CDR. This patch
+ sets KEEP_VARS flag on the most recent CDR on the stack (the CDR
+ used for forking). (closes issue ASTERISK-23260) Reported by:
+ zvision Patches: app_forkcdr.diff uploaded by zvision (license
+ 5755) ........ Merged revisions 408747 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * contrib/scripts/install_prereq: install_prereq: Missing
+ uuid[-dev] for debian distros Added uuid and uuid-dev to install
+ prereq script. (closes issue ASTERISK-23255) Reported by: Rusty
+ Newton
+
+ * contrib/scripts/install_prereq, main/rtp_engine.c: rtp_engine:
+ Dynamic payload change in rtp mapping not supported Asterisk
+ didn't support the dynamic payload change in rtp mapping in the
+ 200 OK response. Scenario: Asterisk sends the INVITE proposing
+ alaw and telephone-event, it proposes rtpmap:101 for
+ telephone-event. Peer responds with 2xx, it answers with alaw and
+ telephone-event also, but it proposes a different rtpmap number
+ (rtpmap:103) for telephone-event. Expected Behaviour: Asterisk
+ should honour the rtpmapping in the response and send DTMF
+ packets using 103 as payload type for DTMF. Actual Behaviour:
+ Asterisk sends DTMF packets using payload type 101. With this
+ patch asterisk now supports changes that can occur in the rtp
+ mapping in the response. (closes issue ASTERISK-23279) Reported
+ by: NITESH BANSAL Review:
+ https://reviewboard.asterisk.org/r/3225/ Patches:
+ dynamic_payload_change.patch uploaded by nbansal (license 6418)
+
+ * main/rtp_engine.c, /: rtp_engine: Output mixup in
+ ${CHANNEL(rtpqos,audio,all)} Fixed the output of
+ CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
+ (closes issue ASTERISK-23261) Reported by: rsw686 Patches:
+ rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged
+ revisions 408646 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c, main/channel.c: channel.c: MOH is not
+ working for transferee after attended transfer Updated the code
+ to check to see if MOH is playing on the transferor and if so
+ then start it on the channel that replaces it during a
+ masquerade. Example scenario of the problem: Alice calls Bob and
+ then Bob begins the attended transfer process into a queue. Upon
+ going on hold Alice hears music and so does Bob once he is in the
+ queue. Bob then transfers Alice into the queue and then music for
+ Alice stops even though she should be hearing it since has now
+ replaced Bob in the queue. The problem that was occurring is that
+ once the channel was masqueraded the app (queues, confbridge,
+ etc...) had no way of knowing that the channel had just been
+ swapped out thus it did not start music for the present channel.
+ Credit to Olle Johansson for pointing me in the right direction
+ on this issue. (closes issue ASTERISK-19499) Reported by: Timo
+ Teräs Review: https://reviewboard.asterisk.org/r/3226/ ........
+ Merged revisions 408642 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 10:40 +0000 [r408590] Alexandr Anikin <may at telecom-service.ru>
+
+ * /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
+ variables ........ Merged revisions 408589 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 00:47 +0000 [r408537] Michael L. Young <elgueromexicano at gmail.com>
+
+ * /, apps/app_chanspy.c: app_chanspy: Documentation Update To
+ Clarify "x" Option When using the "x" option (specify a DTMF
+ digit to exit the application), it is not obvious in the
+ documentation that this only works when spying on a channel. If a
+ channel being used to spy on other channels is waiting to connect
+ to a channel or is no longer attached to a channel, the DTMF is
+ ignored. As noted on the issue tracker, since there are
+ workarounds available and this is a rarely used option we are
+ opting for a documentation change here. (closes issue
+ ASTERISK-22661) Reported by: Chris Hillman Patches:
+ asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
+ Young (license 5026) Review:
+ https://reviewboard.asterisk.org/r/2990/ ........ Merged
+ revisions 408536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-20 02:41 +0000 [r408448] Rusty Newton <rnewton at digium.com>
+
+ * apps/app_queue.c, /: apps/app_queue - Fix incorrect Macro
+ parameter documentation Macro is executed on the called channel,
+ not the calling channel. (closes issue ASTERISK-23069) Reported
+ By: Bryan Anderson ........ Merged revisions 408447 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-19 19:05 +0000 [r408388] Richard Mudgett <rmudgett at digium.com>
+
+ * main/config.c, /: config: Add file size and nanosecond resolution
+ fields to the cached modified config file information. Repeatedly
+ modifying config files and reloading too fast sometimes fails to
+ reload the configuration because the cached modification
+ timestamp has one second resolution. * Added file size and
+ nanosecond resolution fields to the cached config file
+ modification timestamp information. Now if the file size changes
+ or the file system supports nanosecond resolution the modified
+ file has a better chance of being detected for reload. * Added a
+ missing unlock in an off-nominal code path. (closes issue
+ AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
+ ........ Merged revisions 408387 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-19 11:45 +0000 [r408312-408330] Alexandr Anikin <may at telecom-service.ru>
+
+ * addons/ooh323c/src/ooCapability.c, /,
+ addons/ooh323c/src/ooh245.c: process receiveAndTransmit user
+ input remote caps instead of receive only send receiveAndTransmit
+ user input our caps instead of receive only ........ Merged
+ revisions 408328 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * addons/ooh323c/src/ooh323.c: Allow different socket and
+ signalling ip on h.323 connection if gk mode is active Reported
+ by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by:
+ Gabriele Odone
+
+2014-02-16 03:15 +0000 [r408193-408201] Matthew Jordan <mjordan at digium.com>
+
+ * main/pbx.c, /: pbx: Handle a completely empty dialplan during a
+ context merge It is highly unlikely, but - at least in Asterisk
+ 12 - theoretically possible to load Asterisk with no dialplan
+ whatsoever. If that occurs, and some other module (that is not a
+ pbx module) attempts to merge its contexts into the dialplan, the
+ existing merge routine will crash. This is because it is not
+ insane, and rightly believes that you provided some sort of
+ dialplan, somewhere. This patch will gracefully merge the
+ contexts in such a case. Note that this is highly unlikely to
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