[asterisk-commits] lmadsen: tag 1.6.1.0-rc2 r180302 - /tags/1.6.1.0-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Mar 4 18:09:53 CST 2009


Author: lmadsen
Date: Wed Mar  4 18:09:49 2009
New Revision: 180302

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=180302
Log:
Importing files for 1.6.1.0-rc2 release

Added:
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    tags/1.6.1.0-rc2/.version   (with props)
    tags/1.6.1.0-rc2/ChangeLog   (with props)

Added: tags/1.6.1.0-rc2/.lastclean
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Added: tags/1.6.1.0-rc2/ChangeLog
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--- tags/1.6.1.0-rc2/ChangeLog (added)
+++ tags/1.6.1.0-rc2/ChangeLog Wed Mar  4 18:09:49 2009
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+2009-03-04  Leif Madsen <lmadsen at digium.com>
+
+	* Released Asterisk 1.6.1.0-rc2
+
+2009-03-04 21:09 +0000 [r180263]  Russell Bryant <russell at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 180261 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r180261 |
+	  russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines
+	  Resolve object matching issues related to the removal of the
+	  sip_user object. Previously, chan_sip had both sip_peer and
+	  sip_user objects in memory. A patch went in to remove sip_user to
+	  simplify the code, since everything could be done with just
+	  sip_peer. This patch resolves some regressions found that were
+	  introduced by those changes. This code comes from
+	  svn/asterisk/team/group/sip-object-matching/. Here is a list of
+	  the changes that have been made: 1) When doing a match by name
+	  with the find_peer() function, make it much easier to specify
+	  which objects should be matched by having a parameter that
+	  specifies exactly which object types should be considered. Also,
+	  update find_by_name() to handle this parameter. Finally, update
+	  all code to use the new option values. 2) When looking up an
+	  object for an outbound request by name, consider peers only.
+	  (create_addr()) 3) Only match peers on an incoming registration
+	  request. 4) When doing authentication (except for SUBSCRIBE),
+	  look up users by name, instead of all objects by name. 5) When
+	  doing authentication (except for SUBSCRIBE), after looking for a
+	  user by name, look for a peer by IP address, instead of all
+	  objects by IP address. 6) When handling the SIP qualify CLI
+	  command or manager action, look for a peer by name, instead of
+	  any object by name. 7) When handling the SIP unregister CLI
+	  command, look for a peer by name, instead of any object by name.
+	  9) In sip_do_debug_peer(), search for a peer by name, instead of
+	  any object by name. 9) When handling the SIPPEER() dialplan
+	  function, search for a peer by name, instead of any object by
+	  name. 10) In the following session timer related functions,
+	  st_get_se(), st_get_refresher(), and st_get_mode(), when looking
+	  for an object for a given sip_pvt using pvt->peername, look for a
+	  peer by name, instead of any object by name. 11) Fix build_peer()
+	  to properly handle the case where separate type=peer and
+	  type=user entries were specified in sip.conf. (closes issue
+	  #14505) Reported by: lmadsen Review:
+	  http://reviewboard.digium.com/r/172/ ........
+
+2009-03-04 19:27 +0000 [r180122-180197]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/callerid.c: Merged revisions 180195 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) |
+	  11 lines Merged revisions 180194 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4
+	  lines Look for the number in a callerid string starting from the
+	  end. This way a value using <> can exist in the name portion.
+	  (issue #AST-194) ........ ................
+
+	* apps/app_dial.c, /: Merged revisions 180120 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 |
+	  file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines
+	  Remove duplicate 'k' and 'K' Dial options. (closes issue #14601)
+	  Reported by: alecdavis Patches: app_dial.optionk.diff.txt
+	  uploaded by alecdavis (license 585) ........
+
+2009-03-03 23:39 +0000 [r180080]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c, include/asterisk/app.h, apps/app_read.c, /,
+	  main/app.c: Merged revisions 180032 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 |
+	  dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines
+	  app_read does not break from prompt loop with user terminated
+	  empty string In app.c, ast_app_getdata is called to stream the
+	  prompts and receive DTMF input. If ast_app_getdata() receives an
+	  empty string caused by the user inputing the end of string
+	  character, in this case '#', it should break from the prompt loop
+	  and return to app_read, but instead it cycles through all the
+	  prompts. I've added a return value for this special case in
+	  ast_readstring() which uses an enum I've delcared in apps.h. This
+	  enum is now used as a return value for ast_app_getdata(). (closes
+	  issue #14279) Reported by: Marquis Patches: fix_app_read.patch
+	  uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded
+	  by dvossel (license 671) Tested by: Marquis, dvossel Review:
+	  http://reviewboard.digium.com/r/177/ ........
+
+2009-03-03 23:31 +0000 [r180077]  Steve Murphy <murf at digium.com>
+
+	* main/ast_expr2.fl, main/ast_expr2.c, utils/Makefile,
+	  utils/expr2.testinput, /, main/ast_expr2.h, main/ast_expr2.y,
+	  main/ast_expr2f.c: Merged revisions 179973 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) |
+	  33 lines Merged revisions 179807 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some
+	  work to do to port these changes to trunk; the check_expr stuff
+	  hasn't been updated here for quite some time, it appears. I added
+	  some more tests to the check_expr2 suite. I had to play around
+	  with the makefile a bit, etc. I added STANDALONE2 #ifdefs to
+	  ast_expr2.y so as not to conflict structure with aelparse.
+	  ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar
+	  2009) | 19 lines These changes allow AEL to better check ${}
+	  constructs within $[...], that are concatenated with text. I
+	  modified and added rules in ast_expr2.fl to better handle the
+	  concatenations. I added some default routines to ast_expr2.y so
+	  the standalone would compile. It also looks like I haven't run
+	  this thru bison since 2.1, so it's good to get this updated. The
+	  Makefile has comments added now for check_expr2 and check_expr to
+	  explain what they are for, and how to run them. The testexpr2s
+	  stuff has been removed, in favor of check_expr2. expr2.testinput
+	  has been updated to include the two expressions that inspired
+	  these changes (from mcnobody on #asterisk this morning) The
+	  regression has been run and all looks well. ........
+	  ................
+
+2009-03-03 22:49 +0000 [r179939-180009]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c, /, configs/queues.conf.sample: Merged revisions
+	  180007 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar
+	  2009) | 22 lines Merged revisions 180006 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar
+	  2009) | 17 lines Clarify some documentation of queues.conf.sample
+	  It had always been possible to explicitly specify a "blank" value
+	  for a sound file in queues.conf and have no sound played back.
+	  The problem with this is that it would result in some ugly CLI
+	  warnings from file.c. This commit introduces a check when playing
+	  a file in app_queue to see if the name of the file is zero-length
+	  and return early if that is the case. Also, the ability to
+	  specify the blank sound files in queues.conf is now mentioned
+	  more clearly in queues.conf.sample (closes issue #14227) Reported
+	  by: caspy ........ ................
+
+	* doc/timing.txt (added), /, res/res_timing_dahdi.c: Merged
+	  revisions 179937 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r179937 |
+	  mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20
+	  lines Add documentation for timing modules used in Asterisk This
+	  document specifies the timing modules available in Asterisk
+	  beginning with Asterisk 1.6.1. The document goes into detail
+	  about the differences between each and gives a general overview
+	  of what timing is used for in Asterisk. There is also a section
+	  which can be used to help customize your setup or to troubleshoot
+	  timing issues you may have. I also added messages to the DAHDI
+	  timing test used in res_timing_dahdi.c that points to this new
+	  documentation if people experience problems. Big thanks to all
+	  who contributed comments on this. (closes issue #14490) Reported
+	  by: mmichelson Patches: timing.txt uploaded by mmichelson
+	  (license 60) Review: http://reviewboard.digium.com/r/164/
+	  ........
+
+2009-03-03 20:09 +0000 [r179905]  Russell Bryant <russell at digium.com>
+
+	* /, apps/app_directed_pickup.c: Merged revisions 179903 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar
+	  2009) | 1 line fix a leaked channel lock (and future deadlock)
+	  when we try to pick up our own channel ........
+
+2009-03-03 18:30 +0000 [r179843]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/features.c: Merged revisions 179841 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) |
+	  16 lines Merged revisions 179840 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9
+	  lines Do not assume that the bridge_cdr is still attached to the
+	  channel when the 'h' exten is finished executing. It is possible
+	  for a masquerade operation to occur when the 'h' exten is
+	  operating. This operation moves the CDR records around causing
+	  the bridge_cdr to no longer exist on the channel where it is
+	  expected to. We can not safely modify it afterwards because of
+	  this, so don't even try. (closes issue #14564) Reported by: meric
+	  ........ ................
+
+2009-03-03 16:48 +0000 [r179744]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 179742 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009)
+	  | 14 lines Merged revisions 179741 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009)
+	  | 6 lines Ensure chan->fdno always gets reset to -1 after
+	  handling a channel fd event. Since setting fdno to -1 had to be
+	  moved, a couple of other code paths that do process an fd event
+	  return early and do not pass through the code path where it was
+	  moved to. So, set it to -1 in a few other places, too. ........
+	  ................
+
+2009-03-03 14:41 +0000 [r179674]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c, /: Merged revisions 179672 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) |
+	  10 lines Merged revisions 179671 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3
+	  lines Move where fdno is set to the default value to *after* the
+	  read callback of the channel driver is called. We have to do this
+	  as the underlying channel driver may need the fdno value to
+	  determine what to read. ........ ................
+
+2009-03-03 13:56 +0000 [r179611]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 179609 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009)
+	  | 17 lines Merged revisions 179608 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009)
+	  | 9 lines Make it easier to detect an improper call to
+	  ast_read(). When you call ast_waitfor() on a channel, the index
+	  into the channel fds array that holds the file descriptor that
+	  poll() determines has input available is stored in fdno. This
+	  patch clears out this value after a call to ast_read() and also
+	  reports errors if ast_read() is called without an fdno set. From
+	  a discussion on the asterisk-dev list. ........ ................
+
+2009-03-03 00:04 +0000 [r179539]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, /: Merged revisions 179537 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009)
+	  | 21 lines Merged revisions 179536 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009)
+	  | 15 lines Fix bridging regression from commit 176701 This fixes
+	  a bad regression where the bridge would exit after an attended
+	  transfer was made. The problem was due to nexteventts getting set
+	  after the masquerade which caused the bridge to return
+	  AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by:
+	  tim_ringenbach ........ ................
+
+2009-03-02 23:39 +0000 [r179535]  Russell Bryant <russell at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 179533 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009)
+	  | 48 lines Merged revisions 179532 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009)
+	  | 40 lines Move ast_waitfor() down to avoid the results of the
+	  API call becoming stale. This call to ast_waitfor() was being
+	  done way too soon in this section of code. Specifically, there
+	  was code in between the call to waitfor and the code that uses
+	  the result that puts the channel in autoservice. By putting the
+	  channel in autoservice, the previous results of ast_waitfor()
+	  become meaningless, as the autoservice thread will do it's own
+	  ast_waitfor() and ast_read() on the channel. So, when we came
+	  back out of autoservice and eventually hit the block of code that
+	  calls ast_read() on the channel, there may not actually be any
+	  input on the channel available. Even though the previous call to
+	  ast_waitfor() in app_meetme said there was input, the autoservice
+	  thread has since serviced the channel for some period of time.
+	  This bug manifested itself while dvossel was doing some testing
+	  of MeetMe in Asterisk trunk. He was using the timerfd timing
+	  module. When the code hit ast_read() erroneously, it determined
+	  that it must have been called because of input on the timer fd,
+	  as chan->fdno was set to AST_TIMING_FD, since that was the cause
+	  of the last legitimate call to ast_read() done by autoservice. In
+	  this test, an IAX2 channel was calling into the MeetMe
+	  conference. It was _much_ more likely to be seen with an IAX2
+	  channel because of the way audio is handled. Every audio frame
+	  that comes in results in a call to ast_queue_frame(), which then
+	  uses ast_timer_enable_continuous() to notify the channel thread
+	  that a frame is waiting to be handled. So, the chances of
+	  ast_waitfor() indicating that a channel needs servicing due to a
+	  timer event on an IAX2 event is very high. Finally, it is
+	  interesting to note that if a different timing interface was
+	  being used, this bug would probably not be noticed. When
+	  ast_read() is called and erroneously thinks that there is a timer
+	  event to handle, it calls the ast_timer_ack() function. The
+	  pthread and dahdi timing modules handle the ack() function being
+	  called when there is no event by simply ignoring it. In the case
+	  of the timerfd module, it results in a read() on the timer fd
+	  that will block forever, as there is no data to read. This caused
+	  Asterisk to lock up very quickly. Thanks to dvossel and
+	  mmichelson for the fun debugging session. :-) ........
+	  ................
+
+2009-03-02 23:12 +0000 [r179471]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/app.c: Merged revisions 179469 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009)
+	  | 17 lines Merged revisions 179468 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009)
+	  | 10 lines When ending a recording with silence detection,
+	  remember to reduce the duration. The end of the recording is
+	  correspondingly trimmed, but the duration was not trimmed by the
+	  number of seconds trimmed, so the saved duration was necessarily
+	  longer than the actual soundfile duration. (closes issue #14406)
+	  Reported by: sasargen Patches: 20090226__bug14406.diff.txt
+	  uploaded by tilghman (license 14) Tested by: sasargen ........
+	  ................
+
+2009-03-02 23:04 +0000 [r179464]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 179462 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009)
+	  | 16 lines Merged revisions 179461 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009)
+	  | 8 lines Ensure that only one thread is calling ast_settimeout()
+	  on a channel at a time. For example, with an IAX2 channel, you
+	  can have both the channel thread and the chan_iax2 processing
+	  threads calling this function, and doing so twice at the same
+	  time is a bad thing. (Found in a debugging session with dvossel
+	  and mmichelson) ........ ................
+
+2009-03-02 20:18 +0000 [r179407]  Jason Parker <jparker at digium.com>
+
+	* /, main/editline/configure, main/editline/np/unvis.c,
+	  main/editline/sys.h, main/editline/configure.in: Merged revisions
+	  179396 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) |
+	  9 lines Merged revisions 179395 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) |
+	  1 line Remove several silly warnings in editline. One about a
+	  broken preprocessor directive, and another about strlcpy/strlcat.
+	  (closes issue #14264) Reported by: dimas ........
+	  ................
+
+2009-03-02 17:19 +0000 [r179362]  Tilghman Lesher <tlesher at digium.com>
+
+	* cdr/cdr_sqlite3_custom.c, /: Merged revisions 179361 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009)
+	  | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not
+	  loaded) ........
+
+2009-03-02 14:14 +0000 [r179293]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/audiohook.c: Merged revisions 179291 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 |
+	  file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix
+	  issue where changing the volume of both directions of audio did
+	  not work. (closes issue #14574) Reported by: KNK Patches:
+	  audiohook_volume_fix.diff uploaded by KNK (license 545) ........
+
+2009-03-01 23:28 +0000 [r179221-179256]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_speech_utils.c, /: Merged revisions 179254 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar
+	  2009) | 5 lines Swap reversed timevals. This was pointed out by
+	  ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........
+
+	* /, channels/chan_sip.c: Merged revisions 179219 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 |
+	  mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18
+	  lines Properly free memory and remove scheduler entries when a
+	  transmission failure occurs. Previously, only the "data" field of
+	  the sip_pkt created during __sip_reliable_xmit was freed when
+	  XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was
+	  called, this inevitably resulted in the reading and writing of
+	  freed memory. XMIT_ERROR is a condition meaning that we don't
+	  want to attempt resending the packet at all. The proper action to
+	  take is to remove the scheduler entry we just created, free the
+	  packet's data as well as the packet itself, and unlink it from
+	  the list of packets on the sip_pvt structure. (closes issue
+	  #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by
+	  mmichelson (license 60) Tested by: Nick_Lewis ........
+
+2009-02-27 21:48 +0000 [r179166]  Russell Bryant <russell at digium.com>
+
+	* configs/ais.conf.sample, res/res_ais.c, /,
+	  doc/distributed_devstate.txt: Merged revisions 179164 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27
+	  Feb 2009) | 2 lines Mark res_ais as experimental, as the binary
+	  event format is subject to change. ........
+
+2009-02-27 21:34 +0000 [r179163]  Tilghman Lesher <tlesher at digium.com>
+
+	* cdr/cdr_sqlite3_custom.c, /: Merged revisions 179161 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009)
+	  | 3 lines If config file is blank, don't load module. (Closes
+	  issue #14563) ........
+
+2009-02-27 21:25 +0000 [r179160]  Russell Bryant <russell at digium.com>
+
+	* /, UPGRADE.txt: Merged revisions 179154 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 |
+	  russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines
+	  Add a note about the ordering of entries in sip.conf in 1.6.1.
+	  ........
+
+2009-02-27 19:06 +0000 [r179059]  Jason Parker <jparker at digium.com>
+
+	* /, doc/tex/channelvariables.tex: Merged revisions 179057 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb
+	  2009) | 8 lines Update documentation for DIALEDTIME and
+	  ANSWEREDTIME variables. (closes issue #14566) Reported by:
+	  klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by
+	  klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by
+	  klaus3000 (license 65) ........
+
+2009-02-27 03:56 +0000 [r178988]  Steve Murphy <murf at digium.com>
+
+	* configs/features.conf.sample, /, main/features.c: Merged
+	  revisions 178986 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) |
+	  26 lines Merged revisions 178956 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 In this
+	  case, it's just a matter of reducing the default timeouts from
+	  2000 to 1000 msec, as the max def feature digit timeout is no
+	  longer halved. ........ r178956 | murf | 2009-02-26 14:27:32
+	  -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default
+	  feature digit timeout to 1000 ms from the previous default of
+	  500. As per bug 14515, a dev discussion arrived at a "mediated
+	  concensus" of a default feature digit timeout of 1.0 sec. Some
+	  voted for 1300; ctooley thought 1500 for distracted phone users
+	  in phone booths; kpfleming put his foot down at 1.0 sec. Users
+	  who found the previous default max delay of 250 msec perfect, are
+	  welcome to override the new default. Notice that I said that 250
+	  msec was the default; wait a minute, you might say, the config
+	  file said it was 500 msec!; well, because of the bug fix for
+	  14515, we found that 500 msec was actually enforcing a max of
+	  250. The bug fix would restore 500 msec, but we felt even that
+	  was a bit tight for most users... 2000 msec was pushed earlier by
+	  mmichelson, so that reduces to 1000 msec after the bug fix.
+	  Enjoy! ........ ................
+
+2009-02-26 17:50 +0000 [r178875]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 178871 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009)
+	  | 6 lines IAX2 prune realtime, minor tweak to last fix A return
+	  statement was missing which caused unexpected cli output. issue
+	  #14479 ........
+
+2009-02-26 17:38 +0000 [r178869]  Steve Murphy <murf at digium.com>
+
+	* /, main/features.c: Merged revisions 178828 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) |
+	  34 lines Merged revisions 178804 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) |
+	  28 lines This patch prevents the feature detection timeout from
+	  being cut in half. Because the ast_channel_bridge() call will
+	  return 0 and pass a frame pointer for both DTMF_BEGIN and
+	  DTMF_END, the feature_timer field in hte config struct is getting
+	  decremented twice, which effectively cuts the digittimeout in
+	  half. I added conditions to the if statement to only let DTMF_END
+	  frames to flow thru, which solved the problem. Also, when the
+	  frame pointer is null, let control flow thru-- this usually
+	  happens on timeouts. I added a comment to the code to explain
+	  what's going on and why. Many thanks to sodom for reporting this
+	  problem. Personnally, it always seemed like something was wrong
+	  with the featuredigittimeout, but I never could quite decide
+	  what... and was too busy to investigate. This bug forced the
+	  issue, and now we know. Sodom had other issues in 14515, but I
+	  couldn't reproduce them. If he still has problems, and wants to
+	  get them solved, he is welcome to reopen 14515. (closes issue
+	  #14515) Reported by: sodom Patches: 14515.patch uploaded by murf
+	  (license 17) Tested by: murf, sodom ........ ................
+
+2009-02-26 16:44 +0000 [r178803]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/file.c: Merged revisions 178801 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 |
+	  file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix
+	  an issue where the timer for file playback would not be stopped
+	  if DAHDI was not installed. (closes issue #14541) Reported by:
+	  grant ........
+
+2009-02-26 16:07 +0000 [r178769]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /: Merged revisions 178767 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009)
+	  | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues.
+	  If "iax2 prune realtime all" was called, it would appear like the
+	  command was successful, but in reality nothing happened. This is
+	  because the reload that was supposed to take place checks the
+	  config files, sees no changes, and does nothing. If there had
+	  been a change in the the config file, the realtime users would
+	  have been marked for deletion and everything would have been
+	  fine. Now prune_users() and prune_peers() are called instead of
+	  reload_config() to prune all users/peers that are realtime. These
+	  functions remove all users/peers with the rtfriend and delme
+	  flags set. iax2_prune_realtime() also lacked the code to properly
+	  delete a single friend. For example. if iax2 prune realtime
+	  <friend> was called, only the peer instance would be removed. The
+	  user would still remain. (closes issue #14479) Reported by:
+	  mousepad99 Review: http://reviewboard.digium.com/r/176/ ........
+
+2009-02-25 12:46 +0000 [r178511]  Russell Bryant <russell at digium.com>
+
+	* main/asterisk.c, /: Merged revisions 178509 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009)
+	  | 10 lines Merged revisions 178508 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009)
+	  | 2 lines Update the copyright year for the main page of the
+	  doxygen documentation. ........ ................
+
+2009-02-24 23:28 +0000 [r178383-178448]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/extensions.conf.sample, /: Merged revisions 178446 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r178446 | tilghman | 2009-02-24 17:27:23 -0600
+	  (Tue, 24 Feb 2009) | 12 lines Merged revisions 178445 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178445 | tilghman | 2009-02-24 17:25:24 -0600 (Tue, 24 Feb 2009)
+	  | 5 lines Add section about the #exec command in configuration
+	  files. (closes issue #14540) Reported by: jtodd Patch by: jtodd,
+	  with additional notes by tilghman (license 14) ........
+	  ................
+
+	* main/asterisk.c, /: Merged revisions 178381 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 |
+	  tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines
+	  Apparently, a void cast doesn't override warn_unused_result.
+	  ........
+
+2009-02-24 20:44 +0000 [r178379-178380]  Russell Bryant <russell at digium.com>
+
+	* Makefile: revert accidental Makefile change.
+
+	* main/rtp.c, Makefile, /: Merged revisions 178374 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r178374 | russell | 2009-02-24 14:39:57 -0600
+	  (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009)
+	  | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset
+	  to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+	  russell ........ ................
+
+2009-02-24 20:41 +0000 [r178305-178377]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, /: Merged revisions 178375 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 |
+	  tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines
+	  The 3 possible errors with pipe(2) are all impossible in this
+	  situation. ........
+
+	* main/asterisk.c, /, utils/astcanary.c: Merged revisions 178342
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24
+	  Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of
+	  depending upon the astcanary process being inherited by init.
+	  ........
+
+	* /, utils/astcanary.c: Merged revisions 178303 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r178303 |
+	  tilghman | 2009-02-24 11:51:36 -0600 (Tue, 24 Feb 2009) | 7 lines
+	  Cause astcanary to exit if Asterisk exits abnormally and doesn't
+	  kill astcanary. Also, add some documentation supporting the use
+	  of astcanary. (closes issue #14538) Reported by: KNK Patches:
+	  asterisk-1.6.x-astcanary.diff uploaded by KNK (license 545)
+	  ........
+
+2009-02-24 15:22 +0000 [r178232]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 178213 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r178213 | file | 2009-02-24 11:18:38 -0400 (Tue, 24 Feb 2009) |
+	  16 lines Merged revisions 178205 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9
+	  lines Skip check for extension when subscribing for MWI. Since
+	  the remote side is not actually subscribing to a specific
+	  extension when subscribing for MWI just skip the check to see if
+	  the extension exists. They can't use it to specify the mailbox
+	  either since we require configuration of that in sip.conf (closes
+	  issue #14531) Reported by: festr ........ ................
+
+2009-02-23 23:22 +0000 [r178172]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c, /: Merged revisions 178142 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009)
+	  | 22 lines Merged revisions 178141 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009)
+	  | 14 lines Fix infinite DTMF when a BEGIN is received without an
+	  END. This commit is related to rev 175124 of 1.4 where a previous
+	  attempt was made to fix this problem. The problem with the
+	  previous patch was that the inserted code needed to go _before_
+	  setting the lastrxts to the current timestamp. Because those were
+	  the same, the dtmfcount variable was never decremented, and so
+	  the END was never sent. In passing, I removed the dtmfsamples
+	  variable which was completed unused. I also removed a redundant
+	  setting of the lastrxts variable. (closes issue #14460) Reported
+	  by: moliveras ........ ................
+
+2009-02-21 16:04 +0000 [r177945]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 177944 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177944 |
+	  tilghman | 2009-02-21 09:59:49 -0600 (Sat, 21 Feb 2009) | 2 lines
+	  On update, test against the existence of sipregs. ........
+
+2009-02-21 12:51 +0000 [r177851]  Michiel van Baak <michiel at vanbaak.info>
+
+	* /, channels/chan_sip.c: Merged revisions 177849 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177849 |
+	  mvanbaak | 2009-02-21 13:22:32 +0100 (Sat, 21 Feb 2009) | 2 lines
+	  make chan_sip.c compile on OpenBSD again. ........
+
+2009-02-20 23:05 +0000 [r177789]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, /: Merged revisions 177787 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009)
+	  | 16 lines Merged revisions 177786 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009)
+	  | 9 lines Don't print the CR-NL combination when we aren't
+	  outputting to the manager. An embedded CR-NL in a CLI command
+	  screws up several AMI parsers that don't expect to see that
+	  combination in the middle of output. (Closes issue #14305)
+	  Reported by: martins Patch by: tilghman ........ ................
+
+2009-02-20 22:27 +0000 [r177785]  Dwayne M. Hubbard <dhubbard at digium.com>
+
+	* /, apps/app_fax.c: Merged revisions 177699 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177699 |
+	  dhubbard | 2009-02-20 14:29:00 -0600 (Fri, 20 Feb 2009) | 9 lines
+	  Make app_fax compatible with spandsp-0.0.6pre4 Prior to
+	  spandsp-0.0.6pre4 the t30_stats_t structure used a
+	  pages_transferred integer to indicate the number of pages
+	  transferred (so far) during the fax session. The
+	  spandsp-0.0.6pre4 release removed the pages_transferred integer
+	  and replaced it with two different integers - pages_tx and
+	  pages_rx. This revision uses the new integers for
+	  spandsp-0.0.6pre4 while maintaining backwards compatibility for
+	  previous spandsp releases. ........
+
+2009-02-20 22:15 +0000 [r177760-177764]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/strings.h: Oops, last merge broke 1.6.1 branch
+
+	* apps/app_system.c, include/asterisk/app.h, /, main/app.c: Merged
+	  revisions 177664 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 |
+	  tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines
+	  Allow semicolons to be escaped, when passing arguments to the
+	  System command. (closes issue #14231) Reported by: jcovert
+	  Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76
+	  (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by
+	  jcovert (license 551) Tested by: jcovert ........
+
+	* include/asterisk/threadstorage.h, /: Merged revisions 177732 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r177732 | tilghman | 2009-02-20 15:25:37 -0600
+	  (Fri, 20 Feb 2009) | 10 lines Merged revisions 177701 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r177701 | tilghman | 2009-02-20 15:15:01 -0600 (Fri, 20 Feb 2009)
+	  | 3 lines This exception does not appear to still be true for
+	  Solaris 10, and OpenSolaris definitely needs it to be removed.
+	  Fixed for snuff-home on -dev channel. ........ ................
+
+2009-02-20 20:34 +0000 [r177700]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, include/asterisk/frame.h: Fixes issue with
+	  undefined audio codecs in chan_iax2 During iax2 call negotiation,
+	  supported codecs are passed in an Information Element containing
+	  a 2 byte field where each bit correlates to a specific codec. In

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