[asterisk-commits] lmadsen: tag 1.4.24-rc1 r180596 - /tags/1.4.24-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 6 12:29:31 CST 2009


Author: lmadsen
Date: Fri Mar  6 12:29:24 2009
New Revision: 180596

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=180596
Log:
Importing files for 1.4.24-rc1 release

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    tags/1.4.24-rc1/ChangeLog   (with props)

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+2009-03-06  Leif Madsen <lmadsen at digium.com>
+
+	* Released 1.4.24-rc1
+
+2009-03-06 18:23 +0000 [r180567]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Make compilation succeed in dev-mode when
+	  IMAP storage is enabled.
+
+2009-03-06 17:19 +0000 [r180532]  David Vossel <dvossel at digium.com>
+
+	* main/enum.c: 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 23:26 +0000 [r180380-180464]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: [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: 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 18:22 +0000 [r180372]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/rtp.c, main/frame.c, include/asterisk/frame.h: 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 19:22 +0000 [r180194]  Joshua Colp <jcolp at digium.com>
+
+	* main/callerid.c: 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)
+
+2009-03-03 23:01 +0000 [r180010]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_dahdi.c: Make sure we still support zapchan in
+	  users.conf, in addition to dahdichan.
+
+2009-03-03 22:48 +0000 [r180006]  Mark Michelson <mmichelson at digium.com>
+
+	* configs/queues.conf.sample, apps/app_queue.c: 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
+
+2009-03-03 18:27 +0000 [r179840]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: 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 18:11 +0000 [r179807]  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: 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 16:45 +0000 [r179741]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: 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 14:38 +0000 [r179671]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c: 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 13:53 +0000 [r179608]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: 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.
+
+2009-03-02 23:54 +0000 [r179536]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: 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 23:34 +0000 [r179532]  Russell Bryant <russell at digium.com>
+
+	* apps/app_meetme.c: 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 23:09 +0000 [r179468]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/app.c: 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 22:58 +0000 [r179461]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: 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 20:14 +0000 [r179395]  Jason Parker <jparker at digium.com>
+
+	* main/editline/configure, main/editline/np/unvis.c,
+	  main/editline/sys.h, main/editline/configure.in: Remove several
+	  silly warnings in editline. One about a broken preprocessor
+	  directive, and another about strlcpy/strlcat. (closes issue
+	  #14264) Reported by: dimas
+
+2009-02-27 19:03 +0000 [r179056]  Jason Parker <jparker at digium.com>
+
+	* doc/channelvariables.txt: 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)
+
+2009-02-26 21:27 +0000 [r178956]  Steve Murphy <murf at digium.com>
+
+	* configs/features.conf.sample, res/res_features.c: 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 17:24 +0000 [r178838]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: IAX2 prune realtime fix 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/
+
+2009-02-26 17:09 +0000 [r178640-178804]  Steve Murphy <murf at digium.com>
+
+	* res/res_features.c: 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
+
+	* main/ast_expr2.fl, main/ast_expr2f.c: This patch completes the
+	  fixes nec. to make 1.4 asterisk dialplan expressions ($[...])
+	  8-bit transparent While I was updating ast_expr2.fl, I missed one
+	  rule that would allow 8-bit chars to be caught in tokens; and in
+	  so doing, it absorbs the ${ sequence and messes up the checking
+	  of raw exprs by AEL. Trunk already has these changes. (closes
+	  issue #14543) Reported by: klaus3000 Patches: patch.14543
+	  uploaded by murf (license 17) Tested by: murf
+
+2009-02-25 12:43 +0000 [r178508]  Russell Bryant <russell at digium.com>
+
+	* main/asterisk.c: Update the copyright year for the main page of
+	  the doxygen documentation.
+
+2009-02-24 23:25 +0000 [r178445]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/extensions.conf.sample: 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)
+
+2009-02-24 20:36 +0000 [r178373]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c: Only set dtmfcount on BEGIN, and ensure it gets reset
+	  to 0 properly. (issue #14460) Reported by: moliveras Tested by:
+	  russell
+
+2009-02-24 17:02 +0000 [r178266]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_dahdiras.c, res/res_musiconhold.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 15:16 +0000 [r178205]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: 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 23:09 +0000 [r178141]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c: 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
+
+2009-02-20 22:59 +0000 [r177701-177786]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c: 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
+
+	* include/asterisk/threadstorage.h: 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 20:17 +0000 [r177696]  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
+	  1.4 only audio codec bits 0-12 are defined, leaving bits 13-15
+	  undefined. By default all bits are enabled unless specified
+	  otherwise. Since its a 2 byte field and 13-15 are not defined,
+	  these bits are never turned off. In trunk, bits 13-15 are
+	  defined, which means 1.4 is advertising support for codecs it
+	  does not have when talking to trunk. I fixed this by adding
+	  #define for undefined audio codec bits. These bits are then
+	  removed from iax2's full bandwidth capabilities. (closes issue
+	  #14283) Reported by: jcovert
+
+2009-02-19 22:51 +0000 [r177540]  Steve Murphy <murf at digium.com>
+
+	* main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This patch
+	  fixes a problem with 8-bit input to the ast_expr2 scanner. The
+	  real culprit was the --full argument to flex in the Makefile!
+	  This causes a 7-bit scanner to be generated. I reviewed the rules
+	  and found one rule where I needed to specifically include 8-bit
+	  chars for a token. I tested against the text supplied by ibercom,
+	  and all looks very well. This has been there a surprisingly long
+	  time! (closes issue #14498) Reported by: ibercom Patches:
+	  14498.patch uploaded by murf (license 17) Tested by: murf
+
+2009-02-19 22:26 +0000 [r177536]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Fix up potential crashes, by reducing the
+	  sharing between interactive and non-interactive threads. (closes
+	  issue #14253) Reported by: Skavin Patches:
+	  20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
+	  Tested by: Skavin
+
+2009-02-19 18:58 +0000 [r177450]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Force a MWI notification after subscribe
+	  request. Reported by the Resiprocate dev team. Thanks!
+
+2009-02-19 16:37 +0000 [r177383]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_speech_utils.c: If we are able to create a speech
+	  structure unset the ERROR variable in case it was previously set.
+	  (issue #LUMENVOX-13)
+
+2009-02-18 22:43 +0000 [r177225]  Steve Murphy <murf at digium.com>
+
+	* pbx/ael/ael.tab.c, pbx/ael/ael.y: This patch fixes a regression
+	  of sorts that was introduced in rev 24425. It basically fixes
+	  AST-190/ABE-1782. What was wrong: the user has 6000 extensions in
+	  one context; and then 6000 contexts, one per extension. The
+	  parser could only handle about 4893 of the 6000 extens in the
+	  single context. This was due to the regression I mentioned. To
+	  get rid of shift/reduce conflicts, Luigi set up right-recursive
+	  lists for globals, context elements, switch lists, and
+	  statements. Right recursive lists got rid of the warnings, but
+	  instead, they use up a tremendous amount of stack space when the
+	  lists are long. I saw this a few years back, and resolved not to
+	  fix it until someone complained. That day has arrived! After the
+	  changes were made, I ran the regression test suite, and there
+	  were no problems. I took the test case the user provided, and
+	  added 100,000 extensions to the single context, that already had
+	  6,000 extens in it. (I'll see your 6, and raise you 100!) It
+	  takes a few minutes to read it all in, check it and generate code
+	  for it, but no problems. So, I think I can say that
+	  fundamentally, there are no longer any limits on the number of
+	  items you can place in contexts, statement blocks, switches, or
+	  globals, beyond your virt mem constraints.
+
+2009-02-18 20:06 +0000 [r177160]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/h323/cisco-h225.cxx, channels/h323/compat_h323.cxx,
+	  autoconf/ast_check_pwlib.m4, channels/h323/cisco-h225.h,
+	  channels/h323/caps_h323.cxx, channels/h323/ast_h323.cxx,
+	  channels/h323/ast_ptlib.h (added), configure,
+	  channels/h323/compat_h323.h, configure.ac,
+	  channels/h323/caps_h323.h, autoconf/ast_prog_sed.m4,
+	  channels/h323/ast_h323.h, channels/h323/chan_h323.h: Modify h323
+	  to build against PTLib as well as the older PWLib Several changes
+	  in PTLib have occurred requiring build time detection. Changes
+	  accounted for include the library name change, config option
+	  change, install location change, and a boolean type change which
+	  is handled by ast_ptlib.h. Also, the sed check has been modified
+	  to properly work with autoconf >= 2.62. (closes issue #14224)
+	  Reported by: bergolth Patches: asterisk-autoconf-sed.patch
+	  uploaded by bergolth (license 661) asterisk-pwlib-v3.patch
+	  uploaded by bergolth (license 661) Tested by: jpeeler
+
+2009-02-18 18:30 +0000 [r177096]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/config.h: Document the return value of the
+	  update method (as requested on -dev list)
+
+2009-02-18 17:41 +0000 [r176945-177039]  Doug Bailey <dbailey at digium.com>
+
+	* main/utils.c: Merged revisions 177035 manually from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 |
+	  dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines
+	  Fixed error where a check for an zero length, terminated string
+	  was needed. ........
+
+	* main/utils.c: Need to take into account the \0 terminator of the
+	  old string to determine the amount available.
+
+2009-02-18 00:34 +0000 [r176810]  Shaun Ruffell <sruffell at digium.com>
+
+	* codecs/codec_dahdi.c: Several changes to codec_dahdi to play nice
+	  with G723. This commit brings in the changes that were living out
+	  on the svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch.
+	  codec_dahdi.c now always uses signed linear as the simple codec
+	  so that a soft g729 codec will not end up being preferred to the
+	  hardware codec. There are also changes to allow codec_dahdi.c to
+	  feed packets to the hardware in the native sample size of the
+	  codec. This solves problems with choppy audio when using G723.
+
+2009-02-17 21:54 +0000 [r176701]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, res/res_features.c, include/asterisk/channel.h:
+	  Modify bridging to properly evaluate DTMF after first warning is
+	  played The main problem is currently if the Dial flag L is used
+	  with a warning sound, DTMF is not evaluated after the first
+	  warning sound. To fix this, a flag has been added in
+	  ast_generic_bridge for playing the warning which ensures that if
+	  a scheduled warning is missed, multiple warrnings are not played
+	  back (due to a feature evaluation or waiting for digits).
+	  ast_channel_bridge was modified to store the nexteventts in the
+	  ast_bridge_config structure as that information was lost every
+	  time ast_channel_bridge was reentered, causing a hangup due to
+	  incorrect time calculations. (closes issue #14315) Reported by:
+	  tim_ringenbach Reviewed on reviewboard:
+	  http://reviewboard.digium.com/r/163/
+
+2009-02-17 21:21 +0000 [r176426-176661]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_local.c: Backport change to 1.4: Prior to
+	  masquerade, move the group definitions to the channel performing
+	  the masq, so that the group count lingers past the bridge.
+	  (closes issue #14275) Reported by: kowalma Patches:
+	  20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
+	  Tested by: kowalma
+
+	* channels/chan_sip.c: After a 'sip reload', qualifies for realtime
+	  peers weren't immediately restarted, instead waiting until the
+	  next registration. We're now caching the qualify across a
+	  reload/restart and starting the qualify immediately upon loading
+	  the peer. (closes issue #14196) Reported by: pdf Patches:
+	  20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
+	  Tested by: pdf
+
+2009-02-16 23:30 +0000 [r176354]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE not
+	  being relayed correctly during bridging This should have been
+	  committed with rev176247, but I missed it. srcupdate frames no
+	  longer break out of the native bridge, but are not being sent to
+	  the other call leg either. This fixs that. issue #13749
+
+2009-02-16 21:41 +0000 [r176254]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/utils.c: correct a logic error in the last stringfields
+	  commit... don't mark additional space as allocated if the string
+	  was built using already-allocated space
+
+2009-02-16 21:39 +0000 [r176249-176252]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_meetme.c: Remove unused variable and make dev-mode
+	  compilation happy
+
+	* apps/app_meetme.c: Open the DAHDI pseudo device and set it to be
+	  nonblocking atomically Apparently on FreeBSD, attempting to set
+	  the O_NONBLOCKING flag separately from opening the file was
+	  causing an "inappropriate ioctl for device" error. While I cannot
+	  fathom why this would be happening, I certainly am not opposed to
+	  making the code a bit more compact/efficient if it also fixes a
+	  bug. (closes issue #14482) Reported by: ys Patches: meetme.patch
+	  uploaded by ys (license 281) Tested by: ys
+
+2009-02-16 21:28 +0000 [r176247]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Fixes issue with AST_CONTROL_SRCUPDATE
+	  breaking out of native bridge In iax2, when a
+	  AST_CONTROL_SRCUPDATE is received it breaks from the native
+	  bridge, but since there is no code path to handle srcupdate it
+	  just goes to be beginning of the loop. This was causing packet
+	  storms of srcupdate frames between servers. Now srcupdate frames
+	  do not break the native bridge for processing. (closes issue
+	  #13749) Reported by: adiemus
+
+2009-02-16 21:10 +0000 [r176216]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/utils.c: fix a flaw in the ast_string_field_build() family
+	  of API calls; these functions made no attempt to reuse the space
+	  already allocated to a field, so every time the field was written
+	  it would allocate new space, leading to what appeared to be a
+	  memory leak.
+
+2009-02-16 15:33 +0000 [r176029]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Don't have the Via header stored as a
+	  stringfield as it can change often during the lifetime of a
+	  dialog. This issue crept up with subscriptions on the AA50. When
+	  an outgoing NOTIFY is sent a new branch value is created and the
+	  Via header is changed to reflect it. Since this was a stringfield
+	  a new spot in the pool was used for the value while the old was
+	  left untouched/unused. If the current pool was full a new pool
+	  was created. This would cause memory usage to increase steadily.
+	  (issue #AA50-2332)
+
+2009-02-15 23:37 +0000 [r175921]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/pbx.c, channels/chan_sip.c, main/devicestate.c,
+	  include/asterisk/manager.h: fix mis-spelling of the word
+	  registered. Reported by De_Mon on #asterisk-dev.
+
+2009-02-15 20:33 +0000 [r175777-175825]  Olle Johansson <oej at edvina.net>
+
+	* formats/format_ilbc.c: format_ilbc does not depend on codec
+	  libraries and can therefore always be made. My mistake. Ursäkta!
+
+	* formats/format_ilbc.c: Disable format_ilbc.so by default, like
+	  codec_ilbc.so
+
+	* channels/chan_sip.c: Make sure that the debug line is not printed
+	  on debug level 0
+
+2009-02-13 21:53 +0000 [r175698]  Jason Parker <jparker at digium.com>
+
+	* include/asterisk/dahdi_compat.h: Zaptel is not DAHDI. Rather,
+	  Zaptel is actually Zaptel. (in case you're confused, DAHDI is
+	  still DAHDI)
+
+2009-02-13 19:47 +0000 [r175407-175590]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Fix a potential crash situation when using
+	  IMAP voicemail If calling into VoiceMailMain when using IMAP
+	  storage, it was possible to crash Asterisk by hanging up the
+	  phone when prompted for a voicemail mailbox. This patch fixes the
+	  issue. While it may appear that this patch is superficial, it
+	  allows code execution to continue to the failure case just below
+	  the IMAP_STORAGE code block where this patch has been applied
+	  (closes issue #14473) Reported by: dwpaul Patches:
+	  voicemail_imap_crash_no_mailbox.patch uploaded by dwpaul (license
+	  689)
+
+	* main/file.c: Fix a place where filestreams were not refcounted
+	  properly This section was already present in trunk and other
+	  branches, but did not exist in 1.4. (closes issue #14395)
+	  Reported by: ZX81 Patches: 14395.patch uploaded by putnopvut
+	  (license 60) Tested by: ZX81
+
+2009-02-12 21:19 +0000 [r175311]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/udptl.c: 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 20:34 +0000 [r175187-175294]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_features.c: 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
+
+	* res/res_features.c: Fix crash in event of failed attempt to
+	  transfer to parking The peer may not necessarily exist, such as
+	  in the case of a transfer to ParkAndAnnounce. In this case don't
+	  try to play a sound to it.
+
+2009-02-12 16:51 +0000 [r175124]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c: Don't send DTMF for infinite time if we do not
+	  receive an END event. I thought that this was going to end up
+	  being a pretty gnarly fix, but it turns out that there was
+	  actually already a configuration option in rtp.conf, dtmftimeout,
+	  that was intended to handle this situation. However, in between
+	  Asterisk 1.2 and Asterisk 1.4, the code that processed the option
+	  got lost. So, this commit brings it back to life. The default
+	  timeout is 3 seconds. However, it is worth noting that having
+	  this be configurable at all is not really the recommended
+	  behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the
+	  time period of extending the tone is necessary to avoid that a
+	  tone "gets stuck". Regardless of the algorithm used, the tone
+	  SHOULD NOT be extended by more than three packet interarrival
+	  times. A slight extension of tone durations and shortening of
+	  pauses is generally harmless. Three seconds will pretty much
+	  _always_ be far more than three packet interarrival times.
+	  However, that behavior is not required, so I'm going to leave it
+	  with our legacy behavior for now. Code from
+	  svn/asterisk/team/russell/issue_14460 (closes issue #14460)
+	  Reported by: moliveras
+
+2009-02-12 10:16 +0000 [r175029]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* channels/chan_gtalk.c: Set the initiator attribute to lowercase
+	  in our replies when receiving calls. This attribute contains a
+	  JID that identifies the initiator of the GoogleTalk voice
+	  session. The GoogleTalk client discards Asterisk's replies if the
+	  initiator attribute contains uppercase characters. (closes issue
+	  #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded
+	  by jcovert (license 551) Tested by: jcovert
+
+2009-02-12 00:19 +0000 [r174997]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c: Revert RTP changes for continuation of DTMF. Proxy
+	  commit by russell via SMS.
+
+2009-02-12 00:01 +0000 [r174985-174986]  Russell Bryant <russell at digium.com>
+
+	* main/rtp.c: Clear out the current event after forcing the end of
+	  a digit
+
+	* main/rtp.c: Fixify infinite DTMF in the case that no RFC2833 END
+	  event is ever received
+
+2009-02-11 20:54 +0000 [r174885]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, apps/app_macro.c: Restore a behavior that was
+	  recently changed, when we fixed issue #13962 and issue #13363
+	  (related to issue #6176). When a hangup occurs during a Macro
+	  execution in earlier 1.4, the h extension would execute within
+	  the Macro context, whereas it was always supposed to execute only
+	  within the main context (where Macro was called). So this fix
+	  checks for an "h" extension in the deepest macro context where a
+	  hangup occurred; if it exists, that "h" extension executes,
+	  otherwise the main context "h" is executed. (closes issue #14122)
+	  Reported by: wetwired Patches: 20090210__bug14122.diff.txt
+	  uploaded by Corydon76 (license 14) Tested by: andrew
+
+2009-02-10 18:50 +0000 [r174644]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Go off hold when we get an empty reinvite
+	  telling us to. (closes issue #14448) Reported by: frawd Patches:
+	  hold_invite_nosdp.patch uploaded by frawd (license 610)
+
+2009-02-10 17:52 +0000 [r174583]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/jitterbuf.c: Improve behavior of jitterbuffer when
+	  maxjitterbuffer is set. This change improves the way the
+	  jitterbuffer handles maxjitterbuffer and dramatically reduces the
+	  number of frames dropped when maxjitterbuffer is exceeded. In the
+	  previous jitterbuffer, when maxjitterbuffer was exceeded, all new
+	  frames were dropped until the jitterbuffer is empty. This change
+	  modifies the code to only drop frames until maxjitterbuffer is no
+	  longer exceeded. Also, previously when maxjitterbuffer was
+	  exceeded, dropped frames were not tracked causing stats for
+	  dropped frames to be incorrect, this change also addresses that
+	  problem. (closes issue #14044) Patches: bug14044-1.diff uploaded
+	  by mnicholson (license 96) Tested by: mnicholson Review:
+	  http://reviewboard.digium.com/r/144/
+
+2009-02-10 02:27 +0000 [r174369]  Steve Murphy <murf at digium.com>
+
+	* apps/app_rpt.c: This patch solves some compiler complaints in
+	  both 32 and 64-bit environments.
+
+2009-02-09 17:11 +0000 [r174282]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Don't do an SRV lookup if a port is
+	  specified RFC 3263 says to do A record lookups on a hostname if a
+	  port has been specified, so that's what we're going to do. See
+	  section 4.2. (closes issue #14419) Reported by: klaus3000
+	  Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by
+	  klaus3000 (license 65)
+
+2009-02-09 14:48 +0000 [r174218]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_musiconhold.c: Don't overwrite our pointer to the music

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