[asterisk-commits] bebuild: tag 10.10.0-digiumphones-rc1 r374688 - /tags/10.10.0-digiumphones-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Oct 8 15:46:57 CDT 2012


Author: bebuild
Date: Mon Oct  8 15:46:55 2012
New Revision: 374688

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=374688
Log:
Importing files for 10.10.0-digiumphones-rc1 release.

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    tags/10.10.0-digiumphones-rc1/.version   (with props)
    tags/10.10.0-digiumphones-rc1/ChangeLog   (with props)

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--- tags/10.10.0-digiumphones-rc1/ChangeLog (added)
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+2012-10-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.10.0-digiumphones-rc1 Released.
+
+2012-10-08 14:19 +0000 [r374656]  Automerge script <automerge at asterisk.org>
+
+	* 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),
+	  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: Merged revisions 374652 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ........ r374652 | mjordan | 2012-10-08 08:46:27 -0500 (Mon, 08
+	  Oct 2012) | 46 lines 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 21:25 +0000 [r374226-374610]  Automerge script <automerge at asterisk.org>
+
+	* main/manager.c, /: Merged revisions 374586 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374586 | dlee | 2012-10-05 15:23:14 -0500 (Fri,
+	  05 Oct 2012) | 34 lines 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
+	  ................
+
+	* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c,
+	  channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged
+	  revisions 374537 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374537 | rmudgett | 2012-10-05 13:25:20 -0500
+	  (Fri, 05 Oct 2012) | 162 lines 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
+	  ................
+
+	* configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions
+	  374476,374481 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374476 | alecdavis | 2012-10-04 15:05:14 -0500
+	  (Thu, 04 Oct 2012) | 13 lines 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
+	  ................ r374481 | alecdavis | 2012-10-04 15:17:16 -0500
+	  (Thu, 04 Oct 2012) | 17 lines 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
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 374457 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374457 | file | 2012-10-04 12:44:38 -0500 (Thu,
+	  04 Oct 2012) | 17 lines 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
+	  ................
+
+	* res/res_agi.c, main/db.c, /: Merged revisions 374427 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374427 | dlee | 2012-10-04 10:37:11 -0500 (Thu,
+	  04 Oct 2012) | 25 lines 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
+	  ................
+
+	* configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions
+	  374385 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374385 | alecdavis | 2012-10-03 23:41:19 -0500
+	  (Wed, 03 Oct 2012) | 36 lines 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, /: Merged revisions 374370 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374370 | alecdavis | 2012-10-03 23:18:44 -0500
+	  (Wed, 03 Oct 2012) | 15 lines _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
+	  ................
+
+	* res/res_jabber.c, /: Merged revisions 374336 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374336 | mjordan | 2012-10-03 21:11:05 -0500
+	  (Wed, 03 Oct 2012) | 31 lines 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: Merged revisions 374300 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r374300 | mjordan | 2012-10-03 12:25:36 -0500 (Wed, 03 Oct 2012)
+	  | 10 lines 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, /: Merged revisions 374231 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374231 | mjordan | 2012-10-02 16:12:30 -0500
+	  (Tue, 02 Oct 2012) | 9 lines 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, /: Merged revisions 374210 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10 ........
+	  r374210 | mjordan | 2012-10-02 12:10:04 -0500 (Tue, 02 Oct 2012)
+	  | 10 lines 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.
+	  ........
+
+2012-10-02 16:41 +0000 [r374208-374209]  Jason Parker <jparker at digium.com>
+
+	* /: Re-enable automerge.
+
+	* 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, main/channel.c, main/format.c,
+	  main/data.c, main/pbx.c, main/manager.c, /, main/ccss.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 ........ Merged
+	  revisions 374178 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-10-02 01:23 +0000 [r374148-374195]  Automerge script <automerge at asterisk.org>
+
+	* /: automerge cancel
+
+	* tests/test_db.c, apps/app_queue.c, main/db.c,
+	  include/asterisk/astdb.h, /: Merged revisions 374132,374135 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374132 | seanbright | 2012-10-01 12:27:22 -0500
+	  (Mon, 01 Oct 2012) | 2 lines Use ast_copy_string instead of
+	  strncpy to guarantee a NUL terminated string. ................
+	  r374135 | seanbright | 2012-10-01 12:52:38 -0500 (Mon, 01 Oct
+	  2012) | 23 lines 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
+	  ................
+
+2012-09-28 19:25 +0000 [r373498-374058]  Automerge script <automerge at asterisk.org>
+
+	* res/res_jabber.c, /: Merged revisions 374045 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r374045 | jrose | 2012-09-28 14:21:10 -0500
+	  (Fri, 28 Sep 2012) | 12 lines 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
+	  ................
+
+	* res/res_agi.c, /: Merged revisions 373990 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373990 | file | 2012-09-28 07:15:48 -0500 (Fri,
+	  28 Sep 2012) | 8 lines 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
+	  ................
+
+	* apps/app_senddtmf.c, /: Merged revisions 373946 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373946 | rmudgett | 2012-09-27 17:12:47 -0500
+	  (Thu, 27 Sep 2012) | 14 lines 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
+	  ................
+
+	* main/loader.c, /: Merged revisions 373910 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373910 | file | 2012-09-27 11:50:46 -0500 (Thu,
+	  27 Sep 2012) | 24 lines 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, /: Merged revisions 373879 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373879 | file | 2012-09-27 06:32:13 -0500 (Thu,
+	  27 Sep 2012) | 14 lines 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
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 373849 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373849 | mmichelson | 2012-09-26 16:11:35 -0500
+	  (Wed, 26 Sep 2012) | 8 lines 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
+	  ................
+
+	* /, apps/app_meetme.c: Merged revisions 373816 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373816 | rmudgett | 2012-09-26 13:15:50 -0500
+	  (Wed, 26 Sep 2012) | 18 lines 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
+	  ................
+
+	* channels/chan_agent.c, configs/agents.conf.sample, /, main/say.c:
+	  Merged revisions 373769,373774 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373769 | mmichelson | 2012-09-25 17:54:13 -0500
+	  (Tue, 25 Sep 2012) | 11 lines 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
+	  ................ r373774 | mmichelson | 2012-09-25 18:08:46 -0500
+	  (Tue, 25 Sep 2012) | 10 lines 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
+	  ................
+
+	* apps/app_voicemail.c, /: Merged revisions 373737 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373737 | mmichelson | 2012-09-25 16:12:40 -0500
+	  (Tue, 25 Sep 2012) | 11 lines 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
+	  ................
+
+	* res/res_rtp_asterisk.c, channels/chan_local.c, /: Merged
+	  revisions 373703,373706 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373703 | kmoore | 2012-09-25 14:34:01 -0500
+	  (Tue, 25 Sep 2012) | 11 lines 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
+	  ................ r373706 | file | 2012-09-25 15:12:02 -0500 (Tue,
+	  25 Sep 2012) | 22 lines 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
+	  ................
+
+	* configs/sip.conf.sample, apps/app_queue.c,
+	  channels/sip/include/sip.h, /, channels/chan_sip.c: Merged
+	  revisions 373665,373675 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373665 | twilson | 2012-09-25 12:35:30 -0500
+	  (Tue, 25 Sep 2012) | 21 lines 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
+	  ................ r373675 | kmoore | 2012-09-25 13:20:04 -0500
+	  (Tue, 25 Sep 2012) | 13 lines "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
+	  ................
+
+	* codecs/ilbc/iLBC_decode.c, codecs/Makefile, /,
+	  channels/chan_sip.c, codecs/ilbc/iLBC_encode.c: Merged revisions
+	  373631,373633,373645 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373631 | jrose | 2012-09-25 11:24:34 -0500
+	  (Tue, 25 Sep 2012) | 10 lines 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
+	  ................ r373633 | rmudgett | 2012-09-25 11:33:31 -0500
+	  (Tue, 25 Sep 2012) | 5 lines 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
+	  ................ r373645 | rmudgett | 2012-09-25 12:19:52 -0500
+	  (Tue, 25 Sep 2012) | 14 lines 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
+	  ................
+
+	* configs/res_odbc.conf.sample, /: Merged revisions 373579 via
+	  svnmerge from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373579 | kmoore | 2012-09-25 08:28:20 -0500
+	  (Tue, 25 Sep 2012) | 11 lines 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
+	  ................
+
+	* res/res_rtp_multicast.c, /: Merged revisions 373551 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373551 | file | 2012-09-25 07:00:23 -0500 (Tue,
+	  25 Sep 2012) | 15 lines 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: Merged revisions 373533 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373533 | file | 2012-09-24 19:11:28 -0500 (Mon,
+	  24 Sep 2012) | 5 lines Add missing checks that I neglected. The
+	  SIP technology and SIP info technology should be considered
+	  equal. ........ Merged revisions 373532 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ................
+
+	* res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+	  373501,373505 via svnmerge from
+	  file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373501 | rmudgett | 2012-09-24 17:11:01 -0500
+	  (Mon, 24 Sep 2012) | 18 lines 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
+	  ................ r373505 | mjordan | 2012-09-24 17:17:02 -0500
+	  (Mon, 24 Sep 2012) | 19 lines 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
+	  ................
+
+	* apps/app_mixmonitor.c, funcs/func_audiohookinherit.c, /,
+	  channels/chan_sip.c: Merged revisions 373466,373468 via svnmerge
+	  from file:///srv/subversion/repos/asterisk/branches/10
+	  ................ r373466 | rmudgett | 2012-09-24 15:44:27 -0500
+	  (Mon, 24 Sep 2012) | 33 lines 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

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