[asterisk-commits] bebuild: tag 10.10.0-rc1 r374683 - /tags/10.10.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Oct 8 15:44:48 CDT 2012
Author: bebuild
Date: Mon Oct 8 15:44:44 2012
New Revision: 374683
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=374683
Log:
Importing files for 10.10.0-rc1 release.
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tags/10.10.0-rc1/.version (with props)
tags/10.10.0-rc1/ChangeLog (with props)
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+2012-10-08 Asterisk Development Team <asteriskteam at digium.com>
+
+ * Asterisk 10.10.0-rc1 Released.
+
+2012-10-08 13:46 +0000 [r374652] Matthew Jordan <mjordan at digium.com>
+
+ * apps/confbridge/conf_state.c (added),
+ apps/confbridge/conf_state_single.c (added),
+ apps/confbridge/conf_state_inactive.c (added),
+ apps/confbridge/conf_state_single_marked.c (added),
+ apps/confbridge/include/confbridge.h,
+ apps/confbridge/include/conf_state.h (added),
+ apps/confbridge/conf_state_multi.c (added),
+ apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c
+ (added), apps/confbridge/conf_state_empty.c (added): Resolve
+ issues in ConfBridge regarding marked, waitmarked, and unmarked
+ users Thank's to Neil Tallim (flan)'s tireless testing, issue
+ reporting, and patches it became clear that app_confbridge had
+ some complex logic in how it handled interactions between marked,
+ waitmarked, and unmarked users. In particular, there were some
+ areas in which the interactions between the users resulted in
+ inconsistent behavior, and app_confbridge was missing logic in
+ how to handle some corner cases. Some areas included: * Poor
+ handling of mixing unmarked and waitmarked users *
+ Inconsistencies in how MOH and muting was applied to various
+ users * Handling of various announcements for different user
+ profile options flan's patches seem to fix the various issues,
+ but highlighted how hard the code could be to maintain. In an
+ attempt to make things easier to maintain and to more fully
+ enumerate the various cases that exist, this patch breaks up the
+ logic into a state machine-like setup. Please note that the
+ various state transitioned are documented on the Asterisk wiki:
+ https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes
+ Review: //https://reviewboard.asterisk.org/r/2072/ Note that for
+ the following issues, mjordan uploaded the patch, although it was
+ written by twilson. Any contributor license discrepency is due to
+ that. (closes issue ASTERISK-19562) Reported by: flan Tested by:
+ flan, mjordan, jrose patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-19726) Reported by:
+ flan Tested by: flan patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283) (closes issue ASTERISK-20181) Reported by:
+ Jonathan White Tested by: Jonathan White patches:
+ bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by
+ twilson (license 6283)
+
+2012-10-05 20:23 +0000 [r374586] dlee <dlee at localhost>:
+
+ * main/manager.c, /: Multiple revisions 374570,374581 ........
+ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) |
+ 22 lines Improve AMI long line error handling In AMI's parser,
+ when it receives a long line (> 1024 characters), it discards
+ that line, but continues to process the message normally.
+ Typically, this is not a problem because a) who has lines that
+ long and b) usually a discarded line results in an invalid
+ message. But if that line is specifying an optional field, then
+ the message will be processed, you get a 'Response: Success', but
+ things don't work the way you expected them to. This patch
+ changes the behavior when a line-too-long parse error occurs. *
+ Changes the log message to avoid way-too-long (and truncated
+ anyways) log messages * Adds a 'parsing' status flag to Response:
+ Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line
+ is too long * Responds with an appropriate error if parsing !=
+ MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow
+ Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581
+ | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line
+ I've committed too much. Reverting part of r374570. ........
+ Merged revisions 374570,374581 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-05 18:25 +0000 [r374537] Richard Mudgett <rmudgett at digium.com>
+
+ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /,
+ channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
+ Merged revisions 374515-374535 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500
+ (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode *
+ Made setup_bc() static. Patches: patch1_unused-code.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882
+ ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500
+ (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan
+ states Patches: patch2_unused-states.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012)
+ | 16 lines chan_misdn: Remove unnecessary null pointer checks and
+ checks for stack->nt * cleanup_bc() is always called with valid
+ bc (or it would've crashed before). * Value of stack->nt is known
+ in advance at some places. * Rename handle_event() to
+ handle_event_te(), handle_frm() to handle_frm_te(). Patches:
+ patch3_checks.diff (license #6372) patch uploaded by Guenther
+ Kelleter Modified JIRA ABE-2882 ................ r374518 |
+ rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Fix spelling in log messages Patches:
+ patch4_spelling.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374519 | rmudgett |
+ 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines
+ chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after
+ calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is
+ emptied, cleaned and set not in use, although
+ misdn_lib_send_event() already did the same. This is bad. When
+ it's not in use we are not allowed to touch it. * Moved log
+ message in front of the resulting actions and fixed it to match
+ the case. Patches: patch5_bccleanup.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012)
+ | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up
+ etc., really bad stuff. * Fix return codes of cb_events() for
+ EVENT_SETUP to use caller's cleanup mechanisms. * Move
+ cl_queue_chan() call after bearer check. Patches:
+ patch6_leaks.diff (license #6372) patch uploaded by Guenther
+ Kelleter JIRA ABE-2882 ................ r374521 | rmudgett |
+ 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines
+ chan_misdn: We must initialize cause on sending a DISCONNECT. We
+ must initialize cause on sending a DISCONNECT, so it is later
+ correctly indicated to ast_channel in case the answer
+ (RELEASE/RELEASE_COMPLETE) does not include one. Patches:
+ patch7_hangupcause.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374522 |
+ rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Remove unused code for upqueue Patches:
+ patch8_unused-upqueue.diff (license #6372) patch uploaded by
+ Guenther Kelleter JIRA ABE-2882 ................ r374523 |
+ rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines
+ chan_misdn: Improve debugging (port number, messages fixed, dups
+ removed) Patches: patch9_debug.diff (license #6372) patch
+ uploaded by Guenther Kelleter JIRA ABE-2882 ................
+ r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012)
+ | 8 lines chan_misdn: Better debug: we can print_bc_info even if
+ there's no ast leg. Patches: patch10_debug-bc-2.diff (license
+ #6372) patch uploaded by Guenther Kelleter Modified. JIRA
+ ABE-2882 ................ r374534 | rmudgett | 2012-10-05
+ 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn:
+ setup_bc() is called too early for an incoming SETUP on TE. This
+ prevents the B channel from being setup for HDLC mode when
+ requested by the bearer capability and config option hdlc=yes. It
+ violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not
+ connect to the channel until a CONNECT ACKNOWLEDGE message has
+ been received." * Call setup_bc() on receipt of
+ CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for
+ PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by
+ Guenther Kelleter Modified. JIRA ABE-2881 ................
+ r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012)
+ | 2 lines chan_misdn: Remove some more deadcode. ................
+ ........ Merged revisions 374536 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 20:17 +0000 [r374476-374481] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User
+ Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of
+ a recompile, allow values to be adjusted in dsp.conf For binary
+ distributions allows easy adjustment for wobbly GSM calls, and
+ other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and
+ DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by:
+ alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2144/ ........ Merged
+ revisions 374479 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's
+ always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if
+ hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis
+ alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2145/ ........ Merged
+ revisions 374475 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 17:44 +0000 [r374457] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Fix a regression from direct media ACLs
+ where the directrtpsetup option no longer works. A check was
+ added for direct media ACLs that immediately forbid remote
+ bridging if there was no bridged channel. This caused
+ directrtpsetup to no longer function as it needs this information
+ before bridging actually occurs. Logic has now been adjusted so
+ if there is no bridged channel a remote bridge will still be
+ attempted. (closes issue ASTERISK-20511) Reported by: kristoff
+ Review: https://reviewboard.asterisk.org/r/2146/ ........ Merged
+ revisions 374456 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 15:37 +0000 [r374427] dlee <dlee at localhost>:
+
+ * /, res/res_agi.c, main/db.c: Fix DBDelTree error codes for AMI,
+ CLI and AGI The AMI DBDelTree command will return Success/Key
+ tree deleted successfully even if the given key does not exist.
+ The CLI command 'database deltree' had a similar problem, but was
+ saved because it actually responded with '0 database entries
+ removed'. AGI had a slightly different error, where it would
+ return success if the database was unavailable. This came from
+ confusion about the ast_db_deltree retval, which is -1 in the
+ event of a database error, or number of entries deleted
+ (including 0 for deleting nothing). * Changed some poorly named
+ res variables to num_deleted * Specified specific errors when
+ calling ast_db_deltree (database unavailable vs. entry not found
+ vs. success) * Fixed similar bug in AGI database deltree, where
+ 'Database unavailable' results in successful result (closes issue
+ AST-967) Reported by: John Bigelow Review:
+ https://reviewboard.asterisk.org/r/2138/ ........ Merged
+ revisions 374426 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 04:41 +0000 [r374370-374385] Alec L Davis <sivad.a at paradise.net.nz>
+
+ * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User
+ configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values
+ Asterisk's DTMF Specifications are based on AT&T specs, which may
+ not be compatible in other countries. Various countries have
+ different specifications for the maximum power level differences
+ between the DTMF low group and high group of frequencies. Power
+ level difference between frequencies for different
+ Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to
+ 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian
+ = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1
+ (2006-03) Now allow 4 variables to be individually configured in
+ dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T
+ specifications Add's the following variables to dsp.conf
+ ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51
+ ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98
+ (closes issue ASTERISK-20442) Reported by: tbsky Tested by:
+ tbsky,alecdavis alecdavis (license 585) Review
+ https://reviewboard.asterisk.org/r/2141/ ........ Merged
+ revisions 374384 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/dsp.c, /: _dsp_init: bring inline with trunk preparation for
+ clean merge of DTMF TWIST patch No functional changes, just
+ style. alecdavis (license 585) Reported by: Alec Davis Tested by:
+ alecdavis related https://reviewboard.asterisk.org/r/2141
+ ........ Merged revisions 374365 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-04 02:11 +0000 [r374178-374336] Matthew Jordan <mjordan at digium.com>
+
+ * /, res/res_jabber.c: Check for presence of buddy in info/dinfo
+ handlers The res_jabber resource module uses the ASTOBJ library
+ for managing its ref counted objects. After calling
+ ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to
+ the object has to be checked to see if the buddy existed. Prior
+ to this patch, the buddy object was not checked for NULL; with
+ this patch in both aji_client_info_handler and aji_dinfo_handler
+ the pointer is checked before used and, if no buddy object was
+ found, the handlers return an error code. This patch does not
+ take the approach that our JID can be used to log in from another
+ resource. If that approach is desired, an improvement could be
+ made to this patch to create the buddy on the fly. This patch
+ seeks only to prevent Asterisk from crashing. Note that multiple
+ people have proposed patches for this issue; the patch being
+ committed here is based on those. (closes issue ASTERISK-19532)
+ Reported by: Karsten Wemheuer Tested by: Byron Clark patches:
+ fix-jabber uploaded by Karsten Wemheuer (license #5930)
+ xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark
+ (license #6157) (closes issue ASTERISK-19557) Reported by:
+ ulugutz ........ Merged revisions 374335 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/ccss.c: Destroy the generic_monitors container after the
+ core_instances in ccss For each item in core_instances disposed
+ of in the shutdown of ccss, any generic monitor instances
+ referenced by the objects will be removed from generic_monitors
+ during their destruction. Hilarity ensues if generic_monitors no
+ longer exists. Thanks to the Asterisk Test Suite's generic_ccss
+ test for complaining loudly when it ran into this.
+
+ * /, main/asterisk.c: Ensure Shutdown AMI event is still fired
+ during Asterisk shutdown Richard pointed out that having the
+ manager dispose of itself gracefully during shutdown meant that
+ the Shutdown event will no longer get fired. This patch moves the
+ AMI event just prior to running the atexit callbacks. ........
+ Merged revisions 374230 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/message.c: Fix findings from check-in on r374177 Richard
+ pointed out two problems with the check-in from r374177: * The
+ ast_msg_shutdown function declaration doesn't match the prototype
+ in main/message.c. * The ref/alloc function usage in astobj2 (in
+ 11+) can use the ao2_t_* variants of the functions to allow the
+ REF_DEBUG flag to enable/disable their debug counterparts.
+
+ * main/channel.c, main/format.c, main/data.c, main/pbx.c,
+ main/manager.c, /, main/ccss.c, channels/chan_agent.c,
+ main/features.c, main/cel.c, main/format_pref.c,
+ main/indications.c, main/message.c, main/asterisk.c, main/db.c:
+ Fix a variety of ref counting issues This patch resolves a number
+ of ref leaks that occur primarily on Asterisk shutdown. It adds a
+ variety of shutdown routines to core portions of Asterisk such
+ that they can reclaim resources allocate duringd initialization.
+ Review: https://reviewboard.asterisk.org/r/2137 ........ Merged
+ revisions 374177 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-10-01 17:52 +0000 [r374132-374135] Sean Bright <sean at malleable.com>
+
+ * include/asterisk/astdb.h, tests/test_db.c, apps/app_queue.c,
+ main/db.c: app_queue: Support persisting and loading of long
+ member lists. Greenlight in #asterisk brought up that he was
+ receiving an error message "Could not create persistent member
+ string, out of space" when running app_queue in Asterisk 10.
+ dump_queue_members() made an assumption that 8K would be enough
+ to store the generated string, but with queues that have large
+ member lists this is not always the case. This patch removes the
+ limitation and uses ast_str instead of a fixed sized buffer. The
+ complicating factor comes from the fact that ast_db_get requires
+ a buffer and buffer size argument, which doesn't let us pull back
+ more than what we pass in, so I introduced a new
+ ast_db_get_allocated() which returns an ast_strdup()'d copy of
+ the value from astdb. As an aside, I did some testing on the
+ maximum size of data that we can store in the BDB library we
+ distribute and was able to store a 10MB string and retrieve it
+ with no problems, so I feel this is a safe patch. Review:
+ https://reviewboard.asterisk.org/r/2136/ ........ Merged
+ revisions 374108 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * main/db.c: Use ast_copy_string instead of strncpy to guarantee a
+ NUL terminated string.
+
+2012-09-28 19:21 +0000 [r374045] Jonathan Rose <jrose at digium.com>
+
+ * /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test'
+ The opinion of development was that it is both improper to have
+ Matt's personal email address used in the source and that the
+ command wouldn't be useful without it. (closes issue AST-467)
+ Reported by: Malcolm Davenport ........ Merged revisions 374032
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-28 12:15 +0000 [r373990] Joshua Colp <jcolp at digium.com>
+
+ * /, res/res_agi.c: Update documentation to make it explicit that
+ "stream file" will not restart musiconhold. (issue
+ ASTERISK-17367) Reported by: oej ........ Merged revisions 373989
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-27 22:12 +0000 [r373946] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference
+ leak using channel name parameter. The SendDTMF channel name
+ parameter has two issues. 1) Crashes if the channel name does not
+ exist. 2) Leaks a channel reference if the channel is the current
+ channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF
+ documentation. * Renamed app to senddtmf_name and tweaked the
+ type. ........ Merged revisions 373945 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-27 16:50 +0000 [r373879-373910] Joshua Colp <jcolp at digium.com>
+
+ * main/loader.c, /: loader: Ensure dependent modules are properly
+ initialized. If an Asterisk module specifies a dependency in
+ ast_module_info.nonoptreq, it is possible for Asterisk to skip
+ calling the modules's .load function. Asterisk was loading and
+ linking the module via load_dynamic_module() but was not adding
+ the module to the resource_heap. Therefore the module was not
+ initialized based on it's priority along with the other modules
+ in the heap. Now use load_resource() instead of
+ load_dynamic_module() for non-optional requirement. This will add
+ the module to the resource_heap so the module can be properly
+ initialized in the correct order. This is required if there are
+ any module global data structures initialized in the .load()
+ callback for the module on platforms which do not support weak
+ references. (issue ASTERISK-20439) Reported by: sruffell Patches:
+ 0001-loader-Ensure-dependent-modules-are-properly-initial.patch
+ uploaded by sruffell (license 5417) ........ Merged revisions
+ 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * channels/chan_local.c, /: Fix an issue where Local channels
+ dialed by app_queue are considered in use immediately. The
+ chan_local channel driver returns a device state of in use even
+ if a created Local channel has not yet been dialed. This fix
+ changes the logic to return a state of not in use until the
+ channel itself has been dialed. (closes issue ASTERISK-20390)
+ Reported by: tim_ringenbach Review:
+ https://reviewboard.asterisk.org/r/2116/ ........ Merged
+ revisions 373878 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-26 21:11 +0000 [r373849] Mark Michelson <mmichelson at digium.com>
+
+ * /, channels/chan_sip.c: Move handling of 408 response so there is
+ no misleading warning message. (closes issue ASTERISK-20060)
+ Reported by: Walter Doekes ........ Merged revisions 373848 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-26 18:15 +0000 [r373816] Richard Mudgett <rmudgett at digium.com>
+
+ * /, apps/app_meetme.c: Fixed meetme tab completion and command
+ documentation. * Removed unnecessary case sensitivity in meetme
+ list, lock, unlock, mute, unmute, and kick commands. * Separated
+ meetme lock/unlock, mute/unmute, and kick commands into their own
+ registered commands to simplify tab completion and parameter
+ checking. meetme_lock_cmd(), meetme_mute_cmd(), and
+ meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue
+ AST-1006) Reported by: John Bigelow Tested by: rmudgett ........
+ Merged revisions 373815 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 23:08 +0000 [r373737-373774] Mark Michelson <mmichelson at digium.com>
+
+ * /, main/say.c: Fix saying of date in Dutch. The Dutch say the
+ date before the month. (closes issue ASTERISK-20353) Reported by:
+ Teun Ouwehand ........ Merged revisions 373773 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * configs/agents.conf.sample, /, channels/chan_agent.c: Remove dead
+ code and documentation for nonexistent feature. multiplelogin was
+ removed from chan_agent back in 1.6.0 when AgentCallbackLogin()
+ was removed. (closes issue AST-948) reported by Steve Pitts
+ ........ Merged revisions 373768 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/app_voicemail.c, /: Fix error where improper IMAP greetings
+ would be deleted. (closes issue ASTERISK-20435) Reported by:
+ fhackenberger Patches: asterisk-20435-imap-del-greeting.diff
+ uploaded by Michael L. Young (License #5026) (with suggested
+ modification made by me) ........ Merged revisions 373735 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 20:12 +0000 [r373706] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_local.c, /: Fix T.38 support when used with
+ chan_local in between. Users of the T.38 API can indicate
+ AST_T38_REQUEST_PARMS on a channel to request that the channel
+ indicate a T.38 negotiation with the parameters present on the
+ channel. The return value of this indication is expected to be
+ AST_T38_REQUEST_PARMS upon success but with chan_local involved
+ this could never occur. This fix changes chan_local to always
+ return AST_T38_REQUEST_PARMS for this situation. If the
+ underlying channel technology on the other side does not support
+ T.38 this would have been determined ahead of time using
+ ast_channel_get_t38_state and an indication would not occur.
+ (closes issue ASTERISK-20229) Reported by: wdoekes Patches:
+ ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review:
+ https://reviewboard.asterisk.org/r/2070/ ........ Merged
+ revisions 373705 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 19:34 +0000 [r373675-373703] Kinsey Moore <kmoore at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Fix an issue where media would not
+ flow for situations where the legacy STUN code is in use. The
+ STUN packets should *not* be blocked by strict RTP. (closes issue
+ ASTERISK-20415) Reported-by: Michele Cicciotti Patch-by: Josh
+ Colp (trunk r369817) ........ Merged revisions 373702 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: "show" completion option for "queue"
+ shouldn't appear twice When tab-completing CLI commands starting
+ with "queue", "show" appeared twice in the list due to the way
+ that Asterisk's tab completion functions and the order in which
+ the commands were registered. The registration order has been
+ altered to resolve this issue. (closes issue AST-940)
+ Reported-by: Steve Pitts ........ Merged revisions 373666 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 17:35 +0000 [r373665] Terry Wilson <twilson at digium.com>
+
+ * /, channels/chan_sip.c, configs/sip.conf.sample,
+ channels/sip/include/sip.h: Properly handle UAC/UAS roles for SIP
+ session timers The SIP session timer mechanism contains a
+ mandatory 'refresher' parameter (included in the Session-Expires
+ header) which is used in the session timer offer/answer signaling
+ within a SIP Invite dialog. It looks like asterisk is
+ interpreting the uac resp. uas role only as the initial role of
+ client and server (caller is uac, callee is uas). The standard
+ rfc 4028 however assigns the client role to the ((RE)-Invite)
+ requester, the server role to the ((RE)-Invite) responder. This
+ patch has Asterisk track the actual refresher as "us" or "them"
+ as opposed to relying on just the configured "uas" or "uac"
+ properties. (closes issue AST-922) Reported by: Thomas Airmont
+ Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged
+ revisions 373652 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 17:19 +0000 [r373633-373645] Richard Mudgett <rmudgett at digium.com>
+
+ * /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix
+ valgrind found memcpy issues in codec_ilbc. Valgrind found
+ codec_ilbc using memcpy instead of memmove for overlapping memory
+ blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231)
+ Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license
+ #5674) patch uploaded by Walter Doekes ........ Merged revisions
+ 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * codecs/Makefile, /: Make rebuild GSM, ilbc, or lpc10 codecs if
+ the respective sources change. ........ Merged revisions 373618
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 16:24 +0000 [r373631] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/chan_sip.c: chan_sip: Set Quality of Service for
+ video rtp instance (closes issue ASTERISK-20201) Reported by:
+ ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license
+ 6008) ........ Merged revisions 373617 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 13:28 +0000 [r373579] Kinsey Moore <kmoore at digium.com>
+
+ * configs/res_odbc.conf.sample, /: Fix documentation for default
+ username in res_odbc This was previously stated to be "root", but
+ is actually the name of the context if unspecified. (closes issue
+ ASTERISK-20258) Reported by: Stefan x ........ Merged revisions
+ 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-25 12:00 +0000 [r373533-373551] Joshua Colp <jcolp at digium.com>
+
+ * res/res_rtp_multicast.c, /: Fix an issue where a caller to
+ ast_write on a MulticastRTP channel would determine it failed
+ when in reality it did not. When sending RTP packets via
+ multicast the amount of data sent is stored in a variable and
+ returned from the write function. This is incorrect as any
+ non-zero value returned is considered a failure while a return
+ value of 0 is success. For callers (such as ast_streamfile) that
+ checked the return value they would have considered it a failure
+ when in reality nothing went wrong and it was actually a success.
+ The write function for the multicast RTP engine now returns -1 on
+ failure and 0 on success, as it should. (closes issue
+ ASTERISK-17254) Reported by: wybecom ........ Merged revisions
+ 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, channels/chan_sip.c: Add missing checks that I neglected. The
+ SIP technology and SIP info technology should be considered
+ equal. (closes issue ASTERISK-20409) Reported by: michele
+ cicciotti privatewave ........ Merged revisions 373532 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-24 22:17 +0000 [r373505] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c, /: Revert change to res_rtp_asterisk
+ committed in r373236 (1.8) The change committed in r373236
+ attempted to account for endpoints that increased their RTP
+ timestamp in DTMF end of event re-transmissions. This change
+ attempted to make Asterisk continue to work with endpoints that
+ failed to follow the RFC while maintaining the fix that allowed
+ for out of order DTMF to be handled. Unfortunately, there is no
+ free lunch, and this patch broke any system that sent DTMF
+ immediately after an RTP session was established or when an SSRC
+ is updated. As such, that patch is being reverted for the
+ previous behavior. Endpoints that erroneously increase the RTP
+ timestamp in DTMF end of event packets will not work properly
+ with Asterisk. (issue ASTERISK-20424) ........ Merged revisions
+ 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-24 22:11 +0000 [r373501] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Be consistent, send From: "Anonymous"
+ <sip:anonymous at anonymous.invalid> When setting
+ CALLERID(pres)=unavailable in the dialplan, the From header in
+ the SIP message contains "Anonymous"
+ <sip:Anonymous at anonymous.invalid>. For consistency, Asterisk
+ should use a lowercase a in the userpart of the URI. * Make the
+ From header use a lowercase A in the userpart of the anonymous
+ URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola
+ Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383)
+ patch uploaded by Antti Yrjola ........ Merged revisions 373500
+ from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-24 21:05 +0000 [r373468] Jonathan Rose <jrose at digium.com>
+
+ * funcs/func_audiohookinherit.c, /, apps/app_mixmonitor.c:
+ func_audiohookinherit: Document some missed sources. This patch
+ also mentions that AUDIOHOOK_INHERIT can be used to transfer
+ MixMonitor audiohooks. There is also wiki that addresses
+ audiohooks and the use of AUDIOHOOK_INHERIT at the following
+ link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks
+ (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........
+ Merged revisions 373467 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-24 20:44 +0000 [r373466] Richard Mudgett <rmudgett at digium.com>
+
+ * /, channels/chan_sip.c: Fix potential reentrancy problems in
+ chan_sip. Asterisk v1.8 and later was not as vulnerable to this
+ issue. * Made find_call() lock each private as it processes the
+ found dialogs. (Primary cause of ABE-2876) * Made the other
+ functions that traverse the dialogs container lock each private
+ as it examines them. * Fix race condition in sip_call() if the
+ thread that sent the INVITE is held up long enough for a response
+ to be processed. The p->initid for the INVITE retransmission
+ could be added after it was canceled by the response processing.
+ * Made __sip_destroy() clean up resource pointers after freeing.
+ This is primarily defensive in case someone has a stale private
+ pointer. * Removed redundant memset() in reqprep(). The call to
+ init_req() already does the memset() and is the first reference
+ to req in reqprep(). * Removed useless set of req.method in
+ transmit_invite(). The calls to initreqprep() and reqprep() have
+ to do this because they memset() the req. JIRA ABE-2876
+ .......... Merged -r373423 from
+ https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
+ ........ Merged revisions 373424 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-24 19:17 +0000 [r373440] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Fix a deadlock caused by a race condition
+ between removing a hint and reloading the dialplan and
+ subscribing to the removed hint. If conditions were right it was
+ possible for both the PBX core and chan_sip to deadlock by both
+ having a lock that the other wants. In the case of the PBX core
+ it had the contexts lock and wanted a SIP dialog lock, while in
+ the case of chan_sip it had the SIP dialog lock and wanted the
+ contexts lock. This fix unlocks the SIP dialog before getting the
+ extension state so that the other thread will not block on trying
+ to lock it. Once the extension state is retrieved the SIP dialog
+ is locked again and life carries on. As the SIP dialog is
+ reference counted it is not possible for it to go away after
+ unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins
+ ........ Merged revisions 373438 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-21 19:08 +0000 [r373300-373343] Jonathan Rose <jrose at digium.com>
+
+ * /, channels/iax2-provision.c: iax2-provision: Fix improper return
+ on failed cache retrieval (closes issue ASTERISK-20337) reported
+ by: John Covert Patches: iax2-provision.c.patch uploaded by John
+ Covert (license 5512) ........ Merged revisions 373342 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * /, apps/app_queue.c: app_queue: Make queue reload members and
+ variants of that work Prior to this patch, 'queue reload members'
+ cli command did not work at all. This also affects the manager
+ function 'QueueReload' when supplied with the 'members: yes'
+ field. (closes issue AST-956) Reported by: John Bigelow ........
+ Merged revisions 373298 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-20 19:14 +0000 [r373245] Joshua Colp <jcolp at digium.com>
+
+ * /, apps/app_meetme.c: Fix incorrect MeetME conference bridge
+ reference count decrementing and sometimes premature destruction.
+ When using the 'e' or 'E' option to MeetMe the configured
+ conference bridges are loaded and examined to see if any are
+ empty. If no conference bridges are empty the caller is prompted
+ to enter the number of one. This operation left around a pointer
+ to the last created conference bridge still containing
+ participants. When the caller that was not able to find any empty
+ conference bridge hung up this pointer was disposed of and the
+ reference count of the conference bridge decremented. If there
+ was only a single participant in the conference bridge it was
+ ultimately destroyed prematurely. (closes issue AST-994) Reported
+ by: John Bigelow ........ Merged revisions 373242 from
+ http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-09-20 18:42 +0000 [r373196-373237] Matthew Jordan <mjordan at digium.com>
+
+ * res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF,
+ accomodate increasing timestamps in End events While endpoints
+ should not be changing the source timestamp between DTMF event
+ packets, the fact is there exists those endpoints that do exactly
+ that. To work around this, we absorb timestamps within the
+ expected re-transmit period. Note that this period only affects
+ End of Event packets, so it should not prevent the detection of
+ new DTMF digits that happen to arrive right on top of each other.
+ (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson
+ Tested by: mjordan, Vladimir Mikhelson Review:
+ https://reviewboard.asterisk.org/r/2124 ........ Merged revisions
+ 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+ * apps/confbridge/conf_config_parser.c: Ensure that all ConfBridge
+ sounds can be set using CONFBRIDGE function The CONFBRIDGE
+ function can be used to set the sounds in a ConfBridge bridge
+ profile. Unfortunately, three sounds were missed in the portion
+ of the code that applies the settings passed in from the
+ function: sound_only_one, join, and leave. This patch makes sure
+ that the sounds passed from the function are applied to the
+ bridge profile. (closes issue ASTERISK-20404) Reported by: univ
+ Tested by: mjordan
+
+2012-09-19 17:05 +0000 [r373179] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Fix a regression where direct media was
+ not permitted for calls using SIP INFO DTMF. A change was
+ committed to fix direct media ACL support. This change wrongly
+ assumed that only a single channel technology structure exists
[... 26991 lines stripped ...]
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