[asterisk-commits] bebuild: tag 10.4.0-rc1 r361168 - /tags/10.4.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Apr 4 13:21:03 CDT 2012


Author: bebuild
Date: Wed Apr  4 13:20:57 2012
New Revision: 361168

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=361168
Log:
Importing files for 10.4.0-rc1 release.

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    tags/10.4.0-rc1/ChangeLog   (with props)

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+2012-04-04  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.4.0-rc1 Released.
+
+2012-04-04 16:38 +0000 [r361091-361143]  Jonathan Rose <jrose at digium.com>
+
+	* main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
+	  channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
+	  apps/app_externalivr.c, channels/chan_iax2.c,
+	  res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU
+	  old-style field designator extensions to fix clang warnings
+	  (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
+	  clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
+	  ........ Also add from the patch the portion in res_fax_spandsp
+	  that didn't apply to 1.8 Merged revisions 361142 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
+	  ASTERISK-19540)
+
+	* /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
+	  nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
+	  by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
+	  ........ Merged revisions 361090 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-03 20:08 +0000 [r360993-361041]  Kinsey Moore <kmoore at digium.com>
+
+	* /, apps/app_transfer.c: Fix the display of documentation for
+	  Transfer This came up while fixing documentation generation for
+	  many other cases where the argument separator was not being
+	  displayed properly. Now that it is displayed properly, it shows
+	  up in the wrong place for Transfer since the '/' is only required
+	  if Tech is present. (related to issue ASTERISK-18168) ........
+	  Merged revisions 361040 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
+	  This change prevents Asterisk from sending RTCP receiver reports
+	  during a remote bridge since it is no longer receiving media and
+	  should not be reporting anything. (related to ASTERISK-19366)
+	  ........ Merged revisions 360987 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-30 21:29 +0000 [r360934]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
+	  logger_thread() had an exit path that failed to release the
+	  logmsgs list lock. * Make logger_thread() exit path unlock the
+	  logmsgs list lock. * Made ast_log() not queue any messages to the
+	  logmsgs list if the close_logger_thread flag is set. (issue
+	  ASTERISK-19463) Reported by: Matt Jordan ........ Merged
+	  revisions 360933 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-29 23:33 +0000 [r360863-360885]  Mark Michelson <mmichelson at digium.com>
+
+	* /, main/features.c: Fix potential race condition during call
+	  pickup. Prior to this patch, a connected line update was queued
+	  during call pickup and then an answer frame was queued. The
+	  original caller would presumably then have his connected line
+	  updated and then the call would be answered. In actuality, the
+	  answer frame was not how the call ended up being answered.
+	  Rather, an odd section in app_dial that checks if the called
+	  channel's state is up. The result is that the order of the
+	  connected line update and the answer were variable. In most
+	  cases, this wasn't actually a bad thing. However, if the 'I'
+	  option was passed to dial, the connected line update would be
+	  inhibited. The fix is to queued the connected line after the
+	  answer frame is queued. This way the race in app_dial is between
+	  two conditions resulting in an answer. This way the connected
+	  line update occurs after the answer every time. (closes issue
+	  ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
+	  Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
+	  Mark Michelson (license 5049) ........ Merged revisions 360884
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Improve accuracy of identifying
+	  information sent in dialog-info SIP NOTIFY requests. This change
+	  makes use of connected party information in addition to caller ID
+	  in order to populate local and remote XML elements in the
+	  dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by:
+	  Maciej Krajewski Tested by: Maciej Krajewski Patches:
+	  local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
+	  ........ Merged revisions 360862 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-28 19:20 +0000 [r360717]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_jingle.c, addons/chan_ooh323.c, /,
+	  cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
+	  channels/chan_gtalk.c: Destroy configs when they are no longer
+	  used https://reviewboard.asterisk.org/r/1834/ ........ Merged
+	  revisions 360712 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-27 18:23 +0000 [r360672]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Make a debug message regarding
+	  subscription changes more accurate. I was getting confused during
+	  some testing why Asterisk was saying that a subscription was
+	  being added when it was clearly being removed. This fixes that
+	  confusion. ........ Merged revisions 360625 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-27 14:35 +0000 [r360489-360575]  Jonathan Rose <jrose at digium.com>
+
+	* /, configure: Updates config with bootstrap where I changed
+	  configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
+	  Clark ........ Merged revisions 360574 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, configure.ac: Fix BETTER_BACKTRACES library detection for
+	  Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
+	  Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
+	  Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
+	  uploaded by Bryon Clark (license 6157) ........ Merged revisions
+	  360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-26 18:41 +0000 [r360472-360476]  Paul Belanger <pabelanger at digium.com>
+
+	* /, CHANGES: Update CHANGES for r360471 ........ Merged revisions
+	  360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/dnsmgr.c, /: Increase verbosity level for ast_verb messages
+	  While this does not fix the issue of the CLI being flooded by
+	  'doing dnsmgr_lookup' messages, increasing the verbosity level
+	  above 5 should help minimize it. ........ Merged revisions 360471
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-24 23:47 +0000 [r360358-360414]  Russell Bryant <russell at russellbryant.com>
+
+	* funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
+	  handling code path. ........ Merged revisions 360413 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_page.c: app_page: Fix a memory leak on every Page().
+	  dial_list is a dynamically allocated array that is allocated at
+	  the beginning of Page() based on how many devices will be dialed.
+	  This was never being freed. ........ Merged revisions 360363 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* apps/app_jack.c, /: app_jack: fix datastore memory leak in error
+	  handling path. ........ Merged revisions 360360 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/ast_expr2.c, /, main/ast_expr2.h, res/ael/ael.tab.c,
+	  main/ast_expr2.y, main/ast_expr2f.c, res/ael/ael_lex.c,
+	  res/ael/ael.tab.h: Multiple revisions 360356-360357 ........
+	  r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
+	  | 6 lines expression parser: Fix (theoretical) memory leak. Fix a
+	  memory leak that is very unlikely to actually happen. If a
+	  malloc() succeeded, but the following strdup() failed, the memory
+	  from the original malloc() would be leaked. ........ r360357 |
+	  russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
+	  Rebuild parsers. This is needed to include the last fix to
+	  main/ast_expr2.y. The changes look much bigger as this
+	  regeneration of the code was done with newer versions of flex and
+	  bison. ........ Merged revisions 360356-360357 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-24 00:37 +0000 [r360263-360310]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, /, channels/sig_pri.c: Make number not available
+	  presentation also set screening to network provided. Q.951
+	  indicates that when the presentation indicator is "Number not
+	  available due to interworking" for a number then the screening
+	  indicator field should be "Network provided". * Made
+	  ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
+	  when the presentation is "Number not available due to
+	  interworking". This fix makes Asterisk consistent and it also
+	  makes it consistent with earlier branches as far as this
+	  presentation value is concerned. * Made pri_to_ast_presentation()
+	  and ast_to_pri_presentation() conversions handle the "Number not
+	  available due to interworking" case better in sig_pri.c. This
+	  change is possible because the minimum required libpri version
+	  (v1.4.11) has the necessary defines in libpri.h. ........ Merged
+	  revisions 360309 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Add missing initialization of
+	  update_redirecting in chan_sip.c ........ Merged revisions 360262
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 14:52 +0000 [r360139]  Jonathan Rose <jrose at digium.com>
+
+	* contrib/scripts/install_prereq, /: Update install_prereq script
+	  to include missing GSM library for debian amd move SQLite3.
+	  (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
+	  debian_install_prereq.diff uploaded by Andrew Latham (license
+	  5985) ........ Merged revisions 360138 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 14:21 +0000 [r360098]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* /, configure, configure.ac: Also detect gmime 2.6 Also detect
+	  gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
+	  (License #5035) <tzafrir.cohen at xorcom.com> ........ Merged
+	  revisions 360087 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 13:28 +0000 [r360088]  Matthew Jordan <mjordan at digium.com>
+
+	* /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
+	  on the final response to a re-INVITE When Asterisk detects a
+	  hangup and cannot send a BYE due to a pending INVITE, it sets the
+	  pendingbye flag and waits for the final response to that INVITE.
+	  When the response is received, it transmits the BYE. If, however,
+	  that INVITE request is a pending re-INVITE, it needs to first
+	  send a CANCEL request to terminate the pending re-INVITE. In that
+	  circumstance, Asterisk was, in some scenarios, clearing the
+	  pendingbye flag after processing the CANCEL request and not
+	  checking for a pending BYE when receiving the final 487 response
+	  to the INVITE. This patch ensures that if the pendingbye flag is
+	  set, it is honored regardless of the nature of the INVITE request
+	  currently in flight. (closes issue ASTERISK-19365) Reported by:
+	  Thomas Arimont Tested by: Thomas Arimont Patches:
+	  bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
+	  6283) Review: https://reviewboard.asterisk.org/r/1807 ........
+	  Merged revisions 360086 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-20 20:37 +0000 [r360034]  Kinsey Moore <kmoore at digium.com>
+
+	* /, apps/app_echo.c: Prevent Echo() from relaying control, null,
+	  and modem frames Echo()'s description states that it echoes
+	  audio, video, and DTMF except for # while it actually echoes any
+	  frame that it receives other than DTMF #. This was causing frame
+	  storms in the test suite in some circumstances where Echo() was
+	  attached to both ends of a pair of local channels and control
+	  frames were being periodically generated. Echo()'s behavior and
+	  description have been modifed so that it only echoes media and
+	  non-# DTMF frames. ........ Merged revisions 360033 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-20 18:11 +0000 [r359982]  Sean Bright <sean at malleable.com>
+
+	* channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in
+	  PeerStatus AMI Event. The PeerStatus event for IAX2 channels
+	  currently includes a header named Post which should have been
+	  Port. So include Port along with Post when emitting the event.
+	  We'll remove Post in trunk.
+
+2012-03-20 17:25 +0000 [r359980]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/manager.c, /, include/asterisk/manager.h: Allow AMI action
+	  callback to be reentrant. Fix AMI module reload deadlock
+	  regression from ASTERISK-18479 when it tried to fix the race
+	  between calling an AMI action callback and unregistering that
+	  action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change.
+	  Locking the ao2 object guaranteed that there were no active
+	  callbacks that mattered when ast_manager_unregister() was called.
+	  Unfortunately, this causes the deadlock situation. The patch
+	  stops locking the ao2 object to allow multiple threads to invoke
+	  the callback re-entrantly. There is no way to guarantee a module
+	  unload will not crash because of an active callback. The code
+	  attempts to minimize the chance with the registered flag and the
+	  maximum 5 second delay before ast_manager_unregister() returns.
+	  The trunk version of the patch changes the API to fix the race
+	  condition correctly to prevent the module code from unloading
+	  from memory while an action callback is active. * Don't hold the
+	  lock while calling the AMI action callback. (closes issue
+	  ASTERISK-19487) Reported by: Philippe Lindheimer Review:
+	  https://reviewboard.asterisk.org/r/1818/ Review:
+	  https://reviewboard.asterisk.org/r/1820/ ........ Merged
+	  revisions 359979 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-16 20:20 +0000 [r359898]  Jonathan Rose <jrose at digium.com>
+
+	* /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
+	  channels This patch addresses a bug with chanspy on local
+	  channels which roughly 50% of the time would create a situation
+	  where chanspy can latch onto a zombie channel, keeping the zombie
+	  alive forever and causing the channel doing the spying to never
+	  be able to hang up. (closes issue ASTERISK-19493) Reported by:
+	  lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
+	  Merged revisions 359892 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-16 08:24 +0000 [r359810]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
+	  uint32_t change from Review:
+	  https://reviewboard.asterisk.org/r/1699/ ........ Merged
+	  revisions 359809 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 19:06 +0000 [r359694-359707]  Matthew Jordan <mjordan at digium.com>
+
+	* /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
+	  manager There exists a remotely exploitable stack buffer overflow
+	  in HTTP digest authentication handling in Asterisk. The
+	  particular method in question is only utilized by HTTP AMI. When
+	  parsing the digest information, the length of the string is not
+	  checked when it is copied into temporary buffers allocated on the
+	  stack. This patch fixes this behavior by parsing out pre-defined
+	  key/value pairs and avoiding unnecessary copies to the stack.
+	  (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+	  by: Matt Jordan ........ Merged revisions 359706 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun
+	  in Milliwatt Milliwatt is vulnerable to a remotely exploitable
+	  stack overrun when using the 'o' option. This occurs due to the
+	  milliwatt_generate function not accounting for
+	  AST_FRIENDLY_OFFSET when calculating the maximum number of
+	  samples it can put in the output buffer. This patch resolves this
+	  issue by taking into account AST_FRIENDLY_OFFSET when determining
+	  the maximum number of samples allowed. Note that at no point is
+	  remote code execution possible. The data that is written into the
+	  buffer is the pre-defined Milliwatt data, and not custom data.
+	  (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
+	  by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
+	  Russell Bryant (license 6283) Note that this patch was written by
+	  Russell, even though Matt uploaded it ........ Merged revisions
+	  359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+	  ........ Merged revisions 359656 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 18:22 +0000 [r359620]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_dial.c, /, apps/app_queue.c: Add missing connected line
+	  macro calls to initial dial for Dial and Queue apps. The
+	  connected line interception macros do not get executed when the
+	  outgoing channel is initially created and that channel's
+	  caller-id is implicitly imported into the incoming channel's
+	  connected line data. If you are using the interception macros,
+	  you would expect that they get run for every change to a
+	  channel's connected line information outside of normal dialplan
+	  execution. Review: https://reviewboard.asterisk.org/r/1817/
+	  ........ Merged revisions 359609 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-15 00:53 +0000 [r359454-359559]  Russell Bryant <russell at russellbryant.com>
+
+	* /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
+	  sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
+	  try_transfer() so that the code isn't (potentially) trying to
+	  read from it while uninitialized. ........ Merged revisions
+	  359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of
+	  uninitialized variable. Avoid potential use of idroster in
+	  gtalk_alloc() before it has been initialized. ........ Merged
+	  revisions 359508 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_chanisavail.c: app_chanisavail: Fix use of
+	  uninitialized variable. Ensure that status is set before it is
+	  used by resetting it during each loop iteration. This could have
+	  resulted in incorrect results from this app. ........ Merged
+	  revisions 359486 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is
+	  initialized. Scan results indicated that this array could be used
+	  uninitialized. At a quick look, it looks correct. In any case,
+	  initializing it is a Good Thing (tm). ........ Merged revisions
+	  359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* include/asterisk/app.h, /: app.h: Always initialize
+	  AST_DECLARE_APP_ARGS(). This patch ensures that the struct
+	  defined by AST_DECLARE_APP_ARGS() is always fully initialized.
+	  I'm not sure if this fixes any real bugs, but it silences a bunch
+	  of warnings from coverity, and is generally a good thing to do
+	  anyway. ........ Merged revisions 359452 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 22:28 +0000 [r359453]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, /, channels/chan_agent.c,
+	  include/asterisk/channel.h: Fix deadlock potential with some
+	  ast_indicate/ast_indicate_data calls. Calling
+	  ast_indicate()/ast_indicate_data() with the channel lock held can
+	  result in a deadlock with a local channel because of how local
+	  channels need to avoid deadlock. ........ Merged revisions 359451
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 17:42 +0000 [r359358]  Matthew Jordan <mjordan at digium.com>
+
+	* /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
+	  missed resynchronizations When a change in time occurs, such that
+	  the timestamps associated with frames being placed into an
+	  adaptive jitter buffer (implemented in jitterbuf.c) are
+	  significantly different then the previously inserted frames, the
+	  jitter buffer checks to see if it needs to be resynched to the
+	  new time frame. If three consecutive packets break the threshold,
+	  the jitter buffer resynchs itself to the new timestamps. This
+	  currently only occurs when history is calculated, and hence only
+	  on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
+	  hand, are never passed to the history calculations. Because of
+	  this, if the jump in time is greater then the maximum allowed
+	  length of the jitter buffer, the JB_TYPE_CONTROL frames are
+	  dropped and no resynchronization occurs. Alterntively, if the
+	  overfill logic is not triggered, the JB_TYPE_CONTROL frame will
+	  be placed into the buffer, but with a time reference that is not
+	  applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
+	  the overflow logic until reads from the jitter buffer reach the
+	  errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
+	  frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
+	  are unlikely to occur in multiples, it perform the
+	  resynchronization on any JB_TYPE_CONTROL frame that breaks the
+	  resynch threshold. Note that this only impacts chan_iax2, as
+	  other consumers of the adaptive jitter buffer use the abstract
+	  jitter buffer API, which does not use JB_TYPE_CONTROL frames.
+	  Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
+	  ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
+	  Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
+	  (license 5722) ........ Merged revisions 359356 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 17:24 +0000 [r359355]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
+	  forked calls generating warnings for voice frames. When connected
+	  line support was added, the wait_for_answer() variable single
+	  changed its meaning slightly. Unfortunately, the places where
+	  single was used did not necessarily get updated to reflect that
+	  change. Also audio/video frames were sent to all forked calls
+	  when the endpoints were never made compatible. * Don't pass
+	  audio/video media frames when the channels have not been made
+	  compatible. * Added handling of AST_CONTROL_SRCCHANGE to
+	  app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
+	  because that frame can also pass a requested MOH class. (closes
+	  issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
+	  ASTERISK-17541) Reported by: clint Review:
+	  https://reviewboard.asterisk.org/r/1805/ ........ Merged
+	  revisions 359344 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-14 10:54 +0000 [r359051-359260]  Russell Bryant <russell at russellbryant.com>
+
+	* include/asterisk/logger.h, /, main/logger.c: Fix bogus
+	  reads/writes of console log levels in asterisk.c This patch
+	  updates the NUMLOGLEVELS define in logger.h to 32, to match the
+	  fact that logger.c implements 32 log levels (because of the
+	  custom log level stuff). asterisk.c uses this define to size an
+	  array of levels per remote console. This array is modified in
+	  ast_console_toggle_loglevel(), which is called by the "logger set
+	  level" CLI command. While the documentation for the CLI command
+	  doesn't make it terribly obvious, you can use this CLI command to
+	  toggle a custom log level on a remote console, as well. However,
+	  doing so led to an invalid array index in asterisk.c. This array
+	  is read from any time a log message is written to a console. So,
+	  all custom log level messages resulted in a bogus read if a
+	  remote console was connected. ........ Merged revisions 359259
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
+	  reads/writes due to incorrect sizeof(). These few places in the
+	  code used sizeof() on h_addr in struct hostent. This is
+	  sizeof(char *). The correct way to get the size of this address
+	  is to use h_length. This error would result in reads/writes of 8
+	  bytes instead of 4 on 64-bit machines. ........ Merged revisions
+	  359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/sched.c, /: Fix inaccurate sizeof() in sched.c. This code
+	  just needed sizeof(int), not sizeof(int *). ........ Merged
+	  revisions 359157 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, utils/astman.c: Fix incorrect sizeof() in astman. ........
+	  Merged revisions 359116 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, res/res_crypto.c: Fix incorrect usage of sizeof() in
+	  res_crypto. In this case, just remove the memset(). There was a
+	  redundant memset that is done correctly just 2 lines later.
+	  ........ Merged revisions 359110 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
+	  ........ Merged revisions 359088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/features.c: Fix incorrect sizeof() usage in features.c.
+	  This didn't actually result in a bug anywhere, luckily. The only
+	  place where the result of these memcpys was used is in app_dial,
+	  and the only field that it read out of ast_call_feature was the
+	  first one, which is an int, so these memcpys always copied just
+	  enough to avoid a problem. ........ Merged revisions 359069 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
+	  ........ Merged revisions 359059 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/pbx.c, /: Don't use a buffer after it goes out of scope. 's'
+	  is set to 'workspace'. Make sure 'workspace' doesn't go out of
+	  scope while the reference to it via 's' is still used. ........
+	  Merged revisions 359056 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h:
+	  Dump cache of published events when a node joins the cluster.
+	  Also use a more reliable method for stopping the poll() thread.
+	  ........ Merged revisions 359053 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
+	  build_tools/menuselect-deps.in, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
+	  apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These
+	  modules are being maintained outside of the tree and have been
+	  for a long time now, so it doesn't make sense to keep them here.
+	  Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged
+	  revisions 359050 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-13 20:36 +0000 [r358944-358989]  Terry Wilson <twilson at digium.com>
+
+	* /, main/features.c: Fix setting CDR variables in the hangup
+	  extension A previous CDR fix for setting CDR variables during a
+	  bridge via custom dialplan features broke setting CDR variables
+	  in the hangup extension. This patch fixes the issue. Review:
+	  https://reviewboard.asterisk.org/r/1794/ ........ Merged
+	  revisions 358978 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* include/asterisk/devicestate.h, /, channels/chan_sip.c,
+	  tests/test_devicestate.c, main/devicestate.c: Make hints for
+	  invalid SIP devices return Unavail, not idle This patch
+	  drastically simplifies the device state aggegation code. The old
+	  method was not only overly complex, but also made it impossible
+	  to return AST_DEVICE_INVALID from the aggregation code. The unit
+	  test update is as a result of fixing that bug. The SIP change
+	  stems from a bug introduced by removing a DNS lookup for
+	  hostname-based SIP channels. (closes issue ASTERISK-16702)
+	  Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged
+	  revisions 358943 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-13 16:58 +0000 [r358811-358860]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed
+	  AEL functionality. ........ Merged revisions 358859 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
+	  utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable
+	  macros in 1.8 to find the next highest "h" extension in a
+	  context, like in 1.4. This change restores functionality that was
+	  present in 1.4, when AEL macros were implemented with the Macro
+	  dialplan application. Macros are fraught with functionality
+	  issues, because they consume a large portion of the underlying
+	  application stack. This limits the ability of AEL users to call
+	  many layers of subroutines, an issue which Gosub does not have
+	  (originally tested to 100,000 levels deep). Therefore, starting
+	  in 1.6.0, AEL macros were implemented with Gosub. However, there
+	  were some implicit behaviors of Macro, which were not replicated
+	  at the same time as with the transition to Gosub, one of which is
+	  documented in the related issue. In particular, the "h" extension
+	  is designed to execute not in the Macro context, but in the
+	  topmost calling context. Due to legacy issues with a misapplied
+	  bugfix many years ago, when a macro exited in 1.4, it looks in
+	  all calling contexts, bubbling up from the deepest level until it
+	  finds an "h" extension. Since AEL hides the complexity of the
+	  underlying dialplan logic from the AEL programmer, it's
+	  reasonable to assume that this behavior should not change in the
+	  transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
+	  break working AEL configurations in the transition to Asterisk
+	  1.8 LTS. This fix is the result, which implements a search for
+	  the "h" extension in all calling Gosub contexts. Fixes
+	  ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
+	  (License #5003) by Tilghman Lesher (with slight modifications for
+	  1.8) Tested by: Johan Wilfer Review:
+	  https://reviewboard.asterisk.org/r/1776/ ........ Merged
+	  revisions 358810 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-08 16:50 +0000 [r358644]  Jonathan Rose <jrose at digium.com>
+
+	* /, channels/chan_sip.c: Make transfer not ignore port information
+	  with SIP. Attempting to transfer with SIP to an address like
+	  1XXXXX at ip.ad.re.ss:5061 would fail because port would be cut from
+	  the host string and ignored. This simply keeps chan_sip from
+	  cutting off the port number during these kinds of transfers.
+	  (closes issue ASTERISK-19321) Reported by: Federico Alves Review:
+	  https://reviewboard.asterisk.org/r/1790/diff/#index_header
+	  ........ Merged revisions 358643 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 18:28 +0000 [r358531]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/sig_ss7.c: Change directly setting _softhangup in
+	  sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
+	  ASTERISK-19372) ........ Merged revisions 358530 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 16:13 +0000 [r358485]  Sean Bright <sean at malleable.com>
+
+	* /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
+	  number of samples set properly. If the wctc4xxp returns more than
+	  a single packet, we need to update the number of samples in the
+	  returned frame accordingly. Acked-by: Shaun Ruffell
+	  <sruffell at digium.com> ........ Merged revisions 358484 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-07 15:17 +0000 [r358436-358441]  Terry Wilson <twilson at digium.com>
+
+	* /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
+	  cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for
+	  ODBC WCHAR fields Without detecting these types, cel_odbc blows
+	  up when the character set for the table is utf8. This also wraps
+	  cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
+	  #ifdef seen in other parts of the code. ........ Merged revisions
+	  358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-06 17:46 +0000 [r358261-358378]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
+	  calls on FXS ports. * Fix referencing the wrong variable in
+	  chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
+	  compiling with -Wshadow and finding this bug. ........ Merged
+	  revisions 358377 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/sig_ss7.c: Drop SS7 call if not connected yet when
+	  INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
+	  clear a failed call as soon as possible. * Made SS7 hangup a call
+	  immediately if it has not connected yet for
+	  INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
+	  inband tone. (closes issue ASTERISK-19372) Reported by: Igor
+	  Nikolaev ........ Merged revisions 358278 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
+	  Setup DSP when SS7 call is connected or early media is available.
+	  Outgoing SS7 calls fail to detect incoming DTMF so any bridged
+	  channel that requires out-of-band DTMF will not work. * Added
+	  sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
+	  The new call converts conditionaled out unconverted code and
+	  shows that the code really did something useful. * Improved some
+	  chan_dahdi DTMF debug messages to help track DTMF handling.
+	  (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........
+	  Merged revisions 358260 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 18:58 +0000 [r358215]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c, /: Eliminate double close of file descriptor in
+	  manager.c The process_output function in manager.c attempted to
+	  call fclose and close immediately afterwards. Since fclose
+	  implies close, this resulted in a potential double free on file
+	  descriptors. This patch changes that behavior and also adds error
+	  checking to fclose and close depending on which was deemed
+	  necessary. Also error messages. Thanks to Rosen Iliev for
+	  pointing out the location of the problem. (closes issue
+	  ASTERISK-18453) Reported By: Jaco Kroon Review:
+	  https://reviewboard.asterisk.org/r/1793/ ........ Merged
+	  revisions 358214 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 16:42 +0000 [r358163]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Defer sending the connected line reinvite
+	  if a reinvite is already in progress. (issue ASTERISK-19355)
+	  Reported by: tomaso (closes issue AST-825) ........ Merged
+	  revisions 358162 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-05 15:59 +0000 [r358116]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
+	  on Replaces errors Asterisk was not setting pendinginvite in the
+	  upper half of handle_request_invite such that the 4xx was
+	  retransmitted repeatedly even though an ack was received for
+	  every retransmission. (closes issue ASTERISK-19303) Patch-by:
+	  Jeremiah Gowdy ........ Merged revisions 358115 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 23:28 +0000 [r357987-358033]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix
+	  unused-but-set-variable warnings All of these were pretty
+	  obviously unused. Some were unused because the code that used
+	  them was #if 0'd. In those cases, I just commented out the
+	  unused-but-set variables. ........ Merged revisions 358029 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c,
+	  channels/misdn/isdn_lib.c: Correct some set-but-unused variable
+	  warnings in the mISDN library. (from kpfleming's commit to trunk
+	  r356292) ........ Merged revisions 358011 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
+	  mode x=++x and x=x=1? Really? ........ Merged revisions 357986
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 21:03 +0000 [r357941]  Kinsey Moore <kmoore at digium.com>
+
+	* /, main/ccss.c, tests/test_event.c, main/event.c,
+	  include/asterisk/strings.h: Fix case-sensitivity for
+	  device-specific event subscriptions and CCSS This change fixes
+	  case-sensitivity for device-specific subscriptions such that the
+	  technology identifier is case-insensitive while the remainder of
+	  the device string is still case-sensitive. This should also
+	  preserve the original case of the device string as passed in to
+	  the event system. CCSS is the only feature affected as it is the
+	  only consumer of device-specific event subscriptions. The second
+	  part of this patch addresses similar case-sensitivity issues

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