[asterisk-commits] lmadsen: tag 1.6.0.7-rc1 r180553 - /tags/1.6.0.7-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 6 12:20:09 CST 2009


Author: lmadsen
Date: Fri Mar  6 12:20:03 2009
New Revision: 180553

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=180553
Log:
Importing files for 1.6.0.7-rc1 release

Added:
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    tags/1.6.0.7-rc1/.version   (with props)
    tags/1.6.0.7-rc1/ChangeLog   (with props)

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--- tags/1.6.0.7-rc1/ChangeLog (added)
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+2009-03-06  Leif Madsen <lmadsen at digium.com>
+
+	* Release 1.6.0.7-rc1
+
+2009-03-06 17:28 +0000 [r180535]  David Vossel <dvossel at digium.com>
+
+	* main/enum.c, /: Merged revisions 180534 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009)
+	  | 15 lines Merged revisions 180532 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009)
+	  | 9 lines Fix handling of backreferences for ENUM lookups enum.c
+	  did not handle regex backtraces correctly. The '\1' in the regex
+	  is a backreference that requires a pattern match to be inserted.
+	  The way the code used to work is that it would find the
+	  backreference and insert the entire input string minus the '+'.
+	  This is incorrect. The regexec() function takes in a variable
+	  called pmatch which is an array of structs containing the start
+	  and end indexes for each backreference substring. The original
+	  code actually passed the pmatch array pointer into regexec but
+	  never did anything with it. Now when a backtrace is found, the
+	  backtrace number is looked up in the pmatch array and the correct
+	  substring is inserted. (closes issue #14576) Reported by:
+	  chris-mac Review: http://reviewboard.digium.com/r/187/ ........
+	  ................
+
+2009-03-05 23:28 +0000 [r180404-180466]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 180465 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600
+	  (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar
+	  2009) | 16 lines [IMAP] Fix message retrieval issues when
+	  identical mailbox names were defined in separate contexts. There
+	  was a fix put in a while back so that an X-Asterisk-VM-Context
+	  message header was added to stored IMAP voicemails. This would
+	  allow for us to differentiate if the same mailbox name was used
+	  in multiple contexts. The problem still left was that not all
+	  places where messages were retrieved actually attempted to use
+	  this header for information when retrieving messages. This commit
+	  fixes that so that MWI and message retrieval from VoiceMailMain
+	  work as expected. (closes issue #13853) Reported by: vicks1
+	  Patches: 13853_v2.patch uploaded by mmichelson (license 60)
+	  Tested by: lmadsen ........ ................
+
+	* apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged
+	  revisions 180383 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar
+	  2009) | 31 lines Merged revisions 180380 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar
+	  2009) | 25 lines Fix broken mailbox parsing when searchcontexts
+	  option is enabled. When using the searchcontexts option in
+	  voicemail.conf, the code made the assumption that all mailbox
+	  names defined were unique across all contexts. However, the code
+	  did nothing to actually enforce this assumption, nor did it do
+	  anything to alert a user that he may have created an ambiguity in
+	  his voicemail.conf file by defining the same mailbox name in
+	  multiple contexts. With this change, we now will issue a nice
+	  long warning if searchcontexts is on and we encounter the same
+	  mailbox name in multiple contexts and ignore any duplicates after
+	  the first box. Whether searchcontexts is enabled or not, if we
+	  come across a duplicate mailbox in the same context, then we will
+	  issue a warning and ignore the duplicated mailbox. I have also
+	  added a small note to voicemail.conf.sample in the explanation
+	  for searchcontexts explaining that you cannot define the same
+	  mailbox in multiple contexts if you have enabled the option.
+	  (closes issue #14599) Reported by: lmadsen Patches: 14599.patch
+	  uploaded by mmichelson (license 60) (with slight modification)
+	  Tested by: lmadsen ........ ................
+
+2009-03-05 18:36 +0000 [r180377]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/rtp.c, main/frame.c, /, include/asterisk/frame.h: Merged
+	  revisions 180373 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar
+	  2009) | 15 lines Merged revisions 180372 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar
+	  2009) | 9 lines Fix problems when RTP packet frame size is
+	  changed During some code analysis, I found that calling
+	  ast_rtp_codec_setpref() on an ast_rtp session does not work as
+	  expected; it does not adjust the smoother that may on the RTP
+	  session, in fact it summarily drops it, even if it has data in
+	  it, even if the current format's framing size has not changed.
+	  This is not good. This patch changes this behavior, so that if
+	  the packetization size for the current format changes, any
+	  existing smoother is safely updated to use the new size, and if
+	  no smoother was present, one is created. A new API call for
+	  smoothers, ast_smoother_reconfigure(), was required to implement
+	  these changes. Review: http://reviewboard.digium.com/r/184/
+	  ........ ................
+
+2009-03-04 19:25 +0000 [r180121-180196]  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 23:35 +0000 [r180078]  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 23:26 +0000 [r180058]  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 [r179971-180008]  Mark Michelson <mmichelson at digium.com>
+
+	* /, configs/queues.conf.sample, apps/app_queue.c: 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 ........ ................
+
+	* apps/app_queue.c: Fix a memory leak when updating a realtime
+	  member field. This was discovered while looking at issue #14353
+
+2009-03-03 18:29 +0000 [r179842]  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 [r179743]  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 14:40 +0000 [r179673]  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 13:55 +0000 [r179610]  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 00:03 +0000 [r179538]  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 23:38 +0000 [r179534]  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 23:15 +0000 [r179473]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 151464 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 |
+	  mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11
+	  lines Make the sip_standard_port function more granular by
+	  allowing separate type and port arguments. This is necessary
+	  because when building our From and Contact headers, we need to be
+	  absolutely sure that we are placing our source port there and not
+	  the peer's source port. (closes issue #12761) Reported by:
+	  asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by
+	  asbestoshead (license 455) ........
+
+2009-03-02 23:11 +0000 [r179470]  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 23:02 +0000 [r179463]  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 20:17 +0000 [r179402]  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 17:58 +0000 [r179360-179363]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_stack.c: KeepAlive application no longer exists, so fix
+	  gosub implementation to not use it. (closes issue #14571)
+	  Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded
+	  by tilghman (license 14) Tested by: tilghman
+
+	* cdr/cdr_sqlite3_custom.c: If cdr registration somehow succeeds
+	  without a config file, don't crash. (closes issue #14563)
+	  Reported by: alerios
+
+2009-03-01 22:07 +0000 [r179220-179222]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Add error checking when updating the "paused"
+	  field of a realtime queue member. This code already existed in
+	  trunk and 1.6.1, but was not in 1.6.0 prior to this commit.
+	  (closes issue #14338) Reported by: fiddur Patches: 14338.patch
+	  uploaded by mmichelson (license 60) Tested by: fiddur
+
+	* /, 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_FAILURE was returned by __sip_xmit. When retrans_pkt was
+	  called, this inevitably resulted in the reading and writing of
+	  freed memory. XMIT_FAILURE 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 21:33 +0000 [r179162]  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 19:05 +0000 [r179058]  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 03:52 +0000 [r178987]  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 [r178874]  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 17:29 +0000 [r178866]  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 16:01 +0000 [r178768]  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 [r178510]  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 [r178382-178447]  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 20:43 +0000 [r178378]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c, /: 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 20:40 +0000 [r178343-178376]  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.
+	  ........
+
+2009-02-24 18:05 +0000 [r178306]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_dahdiras.c: Change include order to make compile on
+	  Centos 5 with DAHDI If BIT_TYPES_DEFINED gets defined before
+	  linux/types.h is included, the __s32 type doesn't get defined
+
+2009-02-24 17:53 +0000 [r178304]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, 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 15:20 +0000 [r178224]  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 23:17 +0000 [r178145]  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 ........ ................
+
+2009-02-23  Leif Madsen <lmadsen at digium.com>
+
+	* Released 1.6.0.6
+
+2009-02-13  Leif Madsen <lmadsen at digium.com>
+
+	* Released 1.6.0.6-rc1 
+
+2009-02-13 16:43 +0000 [r175550]  Joshua Colp <jcolp at digium.com>
+
+	* /, apps/app_record.c: Merged revisions 175549 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175549 |
+	  file | 2009-02-13 12:41:15 -0400 (Fri, 13 Feb 2009) | 4 lines Add
+	  an option to keep the recorded file upon hangup. (closes issue
+	  #14341) Reported by: fnordian ........
+
+2009-02-12 21:41 +0000 [r175369]  Russell Bryant <russell at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 175368 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r175368 |
+	  russell | 2009-02-12 15:41:01 -0600 (Thu, 12 Feb 2009) | 2 lines
+	  Remove useless string copy, and make sscanf safe again ........
+
+2009-02-12 21:27 +0000 [r175347]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/udptl.c, /: Merged revisions 175334 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009)
+	  | 16 lines Merged revisions 175311 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009)
+	  | 9 lines Fix crashes when receiving certain T.38 packets. Also,
+	  increase the maximum size of T.38 packets and warn users when
+	  they try to set the limits above those maximums. (closes issue
+	  #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt
+	  uploaded by Corydon76 (license 14) Tested by: schern ........
+	  ................
+
+2009-02-12 20:59 +0000 [r175299-175301]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/features.c: Fix mistake in merging conflict from 175299.
+
+	* /, main/features.c: Merged revisions 175298 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009)
+	  | 15 lines Merged revisions 175294 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009)
+	  | 9 lines Fix ParkedCall event information for From field in the
+	  case of a blind transfer If the parker information can not be
+	  obtained from the peer, try and see if the BLINDTRANSFER channel
+	  variable has been set. Previously, a blind transfer to the
+	  ParkAndAnnounce app would return nothing for the From. Closes
+	  AST-189 ........ ................
+
+2009-02-12 20:46 +0000 [r175256-175296]  Russell Bryant <russell at digium.com>
+

[... 49388 lines stripped ...]



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