[asterisk-commits] lmadsen: tag 1.4.26-rc2 r199740 - /tags/1.4.26-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jun 9 07:50:55 CDT 2009


Author: lmadsen
Date: Tue Jun  9 07:50:52 2009
New Revision: 199740

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=199740
Log:
Importing files for 1.4.26-rc2 release.

Added:
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    tags/1.4.26-rc2/ChangeLog   (with props)

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+2009-06-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.26-rc2
+
+2009-06-08 19:28 +0000 [r199626-199628]  Sean Bright <sean.bright at gmail.com>
+
+	* include/asterisk/utils.h: Fix a typo in the stack size
+	  calculation just introduced.
+
+	* include/asterisk/utils.h: Increase the size of our thread stack
+	  on 64 bit processors. We were setting the stack size for each
+	  thread to 240KB regardless of architecture, which meant that in
+	  some scenarios we actually had less available stack space on 64
+	  bit processors (pointers use 8 bytes instead of 4). So now we
+	  calculate the stack size we reserve based on the platform's
+	  __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128
+	  bit -> 1008KB (that's right, we're ready for 128 bit processors)
+	  Patch typed by me but written by several members of
+	  #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes
+	  issue #14932) Reported by: jpiszcz Patches:
+	  06052009_issue14932.patch uploaded by seanbright (license 71)
+	  Tested by: seanbright
+
+2009-06-05 21:19 +0000 [r199297]  David Vossel <dvossel at digium.com>
+
+	* main/pbx.c: Fixes issue with hints giving unexpected results.
+	  Hints with two or more devices that include ONHOLD gave
+	  unexpected results. (closes issue #15057) Reported by:
+	  p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel
+	  (license 671) pbx.c.1.4.patch uploaded by p (license 558)
+	  devicestate.c.trunk.patch uploaded by p (license 671) Tested by:
+	  p_lindheimer, dvossel Review:
+	  https://reviewboard.asterisk.org/r/254/
+
+2009-06-04 19:00 +0000 [r199138]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Additional updates to AST-2009-001
+
+2009-06-04 14:14 +0000 [r198957-199022]  Sean Bright <sean.bright at gmail.com>
+
+	* main/asterisk.c, main/loader.c, include/asterisk.h: Safely handle
+	  AMI connections/reload requests that occur during startup. During
+	  asterisk startup, a lock on the list of modules is obtained by
+	  the primary thread while each module is initialized. Issue 13778
+	  pointed out a problem with this approach, however. Because the
+	  AMI is loaded before other modules, it is possible for a module
+	  reload to be issued by a connected client (via Action: Command),
+	  causing a deadlock. The resolution for 13778 was to move
+	  initialization of the manager to happen after the other modules
+	  had already been lodaded. While this fixed this particular issue,
+	  it caused a problem for users (like FreePBX) who call AMI scripts
+	  via an #exec in a configuration file (See issue 15189). The
+	  solution I have come up with is to defer any reload requests that
+	  come in until after the server is fully booted. When a call comes
+	  in to ast_module_reload (from wherever) before we are fully
+	  booted, the request is added to a queue of pending requests. Once
+	  we are done booting up, we then execute these deferred requests
+	  in turn. Note that I have tried to make this a bit more
+	  intelligent in that it will not queue up more than 1 request for
+	  the same module to be reloaded, and if a general reload request
+	  comes in ('module reload') the queue is flushed and we only issue
+	  a single deferred reload for the entire system. As for how this
+	  will impact existing installations - Before 13778, a reload
+	  issued before module initialization was completed would result in
+	  a deadlock. After 13778, you simply couldn't connect to the
+	  manager during startup (which causes problems with
+	  #exec-that-calls-AMI configuration files). I believe this is a
+	  good general purpose solution that won't negatively impact
+	  existing installations. (closes issue #15189) (closes issue
+	  #13778) Reported by: p_lindheimer Patches:
+	  06032009_15189_deferred_reloads.diff uploaded by seanbright
+	  (license 71) Tested by: p_lindheimer, seanbright Review:
+	  https://reviewboard.asterisk.org/r/272/
+
+	* pbx/pbx_spool.c: Fix a possible crash in pbx_spool. We were
+	  trying to reference members of a struct that had previously been
+	  freed. This patch makes sure that we free the struct after it has
+	  been removed from the spooler queue. (closes issue #15072)
+	  Reported by: garlew Patches: spool.diff uploaded by garlew
+	  (license 376)
+
+2009-06-03 15:49 +0000 [r198891]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c, res/res_features.c, include/asterisk/channel.h:
+	  Generic call forward api, ast_call_forward() The function
+	  ast_call_forward() forwards a call to an extension specified in
+	  an ast_channel's call_forward string. After an ast_channel is
+	  called, if the channel's call_forward string is set this function
+	  can be used to forward the call to a new channel and terminate
+	  the original one. I have included this api call in both
+	  channel.c's ast_request_and_dial() and res_feature.c's
+	  feature_request_and_dial(). App_dial and app_queue already
+	  contain call forward logic specific for their application and
+	  options. (closes issue #13630) Reported by: festr Review:
+	  https://reviewboard.asterisk.org/r/271/
+
+2009-06-01 20:07 +0000 [r198665]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_musiconhold.c: If using the old deprecated format, a
+	  reload would cause the class to disappear. (closes issue #14759)
+	  Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt
+	  uploaded by tilghman (license 14) Tested by: lmadsen
+
+2009-05-30 19:36 +0000 [r198370]  Sean Bright <sean.bright at gmail.com>
+
+	* res/res_jabber.c: Properly terminate AMI JabberSend response
+	  messages. The response message (either Error or Success) needs an
+	  extra trailing \r\n after the fields to inform the client that
+	  the message is complete. (closes issue #14876) Reported by: srt
+	  Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright
+	  (license 71) asterisk_14876.patch uploaded by srt (license 378)
+	  trunk-14876-2.diff uploaded by phsultan (license 73)
+
+2009-05-30 03:42 +0000 [r198311]  Russell Bryant <russell at digium.com>
+
+	* res/res_smdi.c: Fix a crash that occurred when MWI SMDI messages
+	  expired. (closes issue #14561) Reported by: cmoss28
+
+2009-05-30 02:46 +0000 [r198251]  Sean Bright <sean.bright at gmail.com>
+
+	* apps/app_dial.c: Treat an empty FORWARD_CONTEXT the same way we
+	  treat a missing one. (closes issue #15056) Reported by:
+	  p_lindheimer Patches: 05292009_bug15056.diff uploaded by
+	  seanbright (license 71) Tested by: p_lindheimer
+
+2009-05-29 18:53 +0000 [r198068]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/cdr.c, main/channel.c, res/res_features.c,
+	  include/asterisk/cdr.h: Use AST_CDR_NOANSWER instead of
+	  AST_CDR_NULL as the default CDR disposition. This change also
+	  involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is
+	  used on originated channels to distinguish: them from dialed
+	  channels. (closes issue #12946) Reported by: meral Patches:
+	  null-cdr2.diff uploaded by mnicholson (license 96) Tested by:
+	  mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested
+	  by: sum
+
+2009-05-29 18:14 +0000 [r197998]  Sean Bright <sean.bright at gmail.com>
+
+	* Makefile: Fix 'make config' target for Slackware. There was a
+	  missing semi-colon after the echo statement in the Makefile that
+	  was causing problems for some users. Fix suggested by reporter.
+	  (closes issue #15225) Reported by: pdavis
+
+2009-05-28 23:57 +0000 [r197895]  Leif Madsen <lmadsen at digium.com>
+
+	* apps/app_mixmonitor.c: Update MixMonitor documentation. Updated
+	  the MixMonitor documentation for the 'b' option so that it is
+	  more obvious that you must not optimize awat the Local channel
+	  when using this option. (issue #14829)
+
+2009-05-28  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.26-rc1
+
+2009-05-28 15:51 +0000 [r197620]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: 'iax show peer blah' now outputs whether or
+	  not peer 'blah' is in trunk mode or not.
+
+2009-05-28 15:27 +0000 [r197588]  Mark Michelson <mmichelson at digium.com>
+
+	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: Allow
+	  for media to arrive from an alternate source when responding to a
+	  reinvite with 491. When we receive a SIP reinvite, it is possible
+	  that we may not be able to process the reinvite immediately since
+	  we have also sent a reinvite out ourselves. The problem is that
+	  whoever sent us the reinvite may have also sent a reinvite out to
+	  another party, and that reinvite may have succeeded. As a result,
+	  even though we are not going to accept the reinvite we just
+	  received, it is important for us to not have problems if we
+	  suddenly start receiving RTP from a new source. The fix for this
+	  is to grab the media source information from the SDP of the
+	  reinvite that we receive. This information is passed to the RTP
+	  layer so that it will know about the alternate source for media.
+	  Review: https://reviewboard.asterisk.org/r/252
+
+2009-05-28 15:21 +0000 [r197562]  Eliel C. Sardanons <eliels at gmail.com>
+
+	* channels/chan_sip.c: Use the address we already know when
+	  reloading a peer with nat=yes. If we already have an address for
+	  a peer, and we are reloading the sip configuration, try to use
+	  that address to contact the peer, instead of getting it from the
+	  Contact. (closes issue #15194) Reported by: ibc Patches:
+	  sip.patch uploaded by eliel (license 64) Tested by: manwe
+
+2009-05-28 14:49 +0000 [r197537]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_chanspy.c, include/asterisk/audiohook.h,
+	  main/audiohook.c: Add flags to chanspy audiohook so that audio
+	  stays in sync. There are two flags being added to the chanspy
+	  audiohook here. One is the pre-existing
+	  AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that
+	  the read and write slinfactories on the audiohook do not skew
+	  beyond a certain tolerance. In addition, there is a new audiohook
+	  flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set,
+	  we do not allow for a slinfactory to build up a substantial
+	  amount of audio before flushing it. For this particular issue,
+	  this means that the person spying on the call will hear the
+	  conversations in real time with very little delay in the audio.
+	  (closes issue #13745) Reported by: geoffs Patches: 13745.patch
+	  uploaded by mmichelson (license 60) Tested by: snblitz
+
+2009-05-28 13:44 +0000 [r197466]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Fix a bug where the flag indicating the
+	  presence of rport would get overwritten by the nat setting. The
+	  presence of rport is now stored as a separate flag. Once the
+	  dialog is setup and authenticated (or it passes through
+	  unauthenticated) the proper nat flag is set. (closes issue
+	  #13823) Reported by: dimas
+
+2009-05-27 20:12 +0000 [r197264]  Sean Bright <sean.bright at gmail.com>
+
+	* Makefile: Use bash explicitly when calling
+	  build_tools/mkpkgconfig from the Makefile. Since we use bashisms
+	  in build_tools/mkpkgconfig, we should call on bash explicitly
+	  when running from the Makefile, otherwise we get errors during a
+	  'make install.' (closes issue #15209) Reported by: seandarcy
+
+2009-05-27 20:07 +0000 [r197259]  Olle Johansson <oej at edvina.net>
+
+	* doc/asterisk-conf.txt: Typo fix
+
+2009-05-27 19:09 +0000 [r197194]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_cut.c: Use a different determinator on whether to
+	  print the delimiter, since leading fields may be blank. (closes
+	  issue #15208) Reported by: ramonpeek Patch by me, though inspired
+	  in part by a patch from ramonpeek
+
+2009-05-27 16:49 +0000 [r197124]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, include/asterisk/channel.h: Fix broken attended
+	  transfers The bridge was terminating immediately after the
+	  attended transfer was completed. The problem was because upon
+	  reentering ast_channel_bridge nexteventts was checked to see if
+	  it was set and if so could possibly return AST_BRIDGE_COMPLETE.
+	  (closes issue #15183) Reported by: andrebarbosa Tested by:
+	  andrebarbosa, tootai, loloski
+
+2009-05-27 13:54 +0000 [r197024]  Sean Bright <sean.bright at gmail.com>
+
+	* apps/app_queue.c: Fix handling of the 'state_interface' option of
+	  the 'queue add member' CLI command. This change relates to
+	  r184980, which was a backport of the state interface changes to
+	  app_queue from trunk. trunk and all of the 1.6.x branches are not
+	  affected. 'queue add member' allows for specifying an interface
+	  to use for device state when adding a queue member via CLI, but
+	  the validation code was not properly updated to reflect this
+	  optional argument. (closes issue #15198) Reported by: loloski
+	  Patches: 05272009_app_queue.diff uploaded by seanbright (license
+	  71) Tested by: loloski
+
+2009-05-26 18:14 +0000 [r196826]  Russell Bryant <russell at digium.com>
+
+	* res/res_convert.c: Resolve a file handle leak. The frames here
+	  should have always been freed. However, out of luck, there was
+	  never any memory leaked. However, after file streams became
+	  reference counted, this code would leak the file stream for the
+	  file being read. (closes issue #15181) Reported by: jkroon
+
+2009-05-26 13:06 +0000 [r196657]  Joshua Colp <jcolp at digium.com>
+
+	* contrib/scripts/safe_asterisk: Remove some bash specific stuff
+	  from safe_asterisk. (closes issue #10812) Reported by: paravoid
+	  Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license
+	  46)
+
+2009-05-22 13:54 +0000 [r196116]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_misdn.c: Fix a bug where using immediate with mISDN
+	  caused a cause code of 16 to get sent back instead of 1 if the
+	  's' extension did not exist. (closes issue #12286) Reported by:
+	  lmamane
+
+2009-05-21 19:04 +0000 [r195991]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Sign problem calculating timestamp for iax
+	  frame leads to no audio on the receiving peer. There are rare
+	  cases in which a frame's delivery timestamp is slightly less than
+	  the iax2_pvt's offset. This causes the pvt's timestamp to be a
+	  small negative number, but since the timestamp value is unsigned
+	  it looks like a huge positive number. This patch checks for this
+	  negative case and sets the ms to zero. A similar check is already
+	  done right below this one in the 'else' statement. (closes issue
+	  #15032) Reported by: guillecabeza Patches:
+	  chan_iax2.c.patch_timestamp uploaded by guillecabeza (license
+	  380) Tested by: guillecabeza (closes issue #14216) Reported by:
+	  Andrey Sofronov
+
+2009-05-21 15:25 +0000 [r195881]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/cdr.c, res/res_features.c, include/asterisk/cdr.h: This
+	  commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and
+	  AST_CDR_FLAG_LOCKED from being updated in certain cases. This is
+	  accomplished by adding two functions to update the answer time
+	  and disposition of calls that checks for the proper lock flags.
+	  These functions are used in the ast_bridge_call() function so
+	  that ForkCDR(A) calls are respected. This patch also modifies the
+	  way ast_bridge_call() chooses the cdr record to base the
+	  bridged_cdr on. Previously the first unlocked cdr record would be
+	  chosen, now instead the first cdr record is chosen and forked cdr
+	  records are moved to the bridge_cdr. This allows the original cdr
+	  record and any forked cdr records to be properly updated with
+	  answer and end times. (closes issue #13797) Reported by: sh0t
+	  Tested by: sh0t (closes issue #14744) Reported by: deepesh
+
+2009-05-21  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.25
+
+2009-05-13  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.25-rc1
+
+2009-05-13 13:38 +0000 [r194208]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c: Fix RFC2833 issues with DTMF getting duplicated and
+	  with duration wrapping over. (closes issue #14815) Reported by:
+	  geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88)
+	  Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue
+	  #14460) Reported by: moliveras Tested by: moliveras
+
+2009-05-13 00:52 +0000 [r194137]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c: Fix logic for how to proceed with a single digit
+	  extension. (closes issue #15091) Reported by: andrew Patches:
+	  20090512__issue15091.diff.txt uploaded by tilghman (license 14)
+	  Tested by: andrew
+
+2009-05-12 22:15 +0000 [r194028]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: This change modifies app_queue to properly
+	  generate CDR records in failure situations. This involves setting
+	  a proper cdr disposition coresponding to the given failure
+	  condition and ensuring the proper information is stored in the
+	  cdr record. (closes issue #13691) Reported by: dferrer Tested by:
+	  mnicholson (closes issue #13637) Reported by: atis Tested by:
+	  atis
+
+2009-05-12 20:39 +0000 [r193955]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Avoid initializing routines if the
+	  authentication fails. Fixes a crash (RR) issue. (closes issue
+	  #14508) Reported by: tiziano Patches:
+	  20090221_2_wrongmailbox.diff.txt uploaded by tiziano (license
+	  377)
+
+2009-05-12 18:18 +0000 [r193880]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Set the invitestate to INV_CANCELLED only if
+	  we are actually sending a SIP CANCEL. The problem was that the
+	  hangup code was setting the invitestate too early. The result of
+	  this was that we would always send a CANCEL request, even if it
+	  was not an appropriate time to do so (e.g. we have not yet
+	  received a provisional response for our INVITE). Note that this
+	  same fix had been applied to trunk and the 1.6.X branches
+	  starting with revision 155467. This is why you will see this
+	  revision being blocked from those places. AST-216
+
+2009-05-11 22:48 +0000 [r193755]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Move 300 bytes around on the stack, to make
+	  more room for an extension buffer. This allows more concurrent
+	  extensions to be copied for a single voicemail, without creating
+	  a possibility of upsetting existing users, where a dialplan could
+	  run out of stack space where it had run fine before.
+	  Alternatively, we could have allocated off the heap, but that is
+	  a larger change and would have increased the chance for
+	  instability introduced by this change. This is really solved
+	  starting in 1.6.0.11, as the use of an ast_str buffer allows an
+	  unlimited number of extensions (up to available memory). We
+	  additionally create a new warning message when the buffer length
+	  is exceeded, permitting administrators to see an issue after the
+	  fact, whereas previously the list was silently truncated. (closes
+	  issue #14739) Reported by: p_lindheimer Patches:
+	  20090417__bug14739.diff.txt uploaded by tilghman (license 14)
+	  Tested by: p_lindheimer
+
+2009-05-11 19:09 +0000 [r193613]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: Sent wrong message to clear a call we
+	  started if the other end has not responed yet. In the state
+	  MISDN_CALLING (i.e. SETUP was sent but no answer has arrived
+	  yet), it is not allowed to clear the call with RELEASE_COMPLETE.
+	  It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only
+	  allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a,
+	  5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer.
+	  JIRA ABE-1862
+
+2009-05-11 17:35 +0000 [r193544]  Leif Madsen <lmadsen at digium.com>
+
+	* funcs/func_channel.c: Document CHANNEL(transfercapability) in CLI
+	  documentation. (issue #15073) Reported by: pkempgen Patches:
+	  20090511__issue15073.diff.txt uploaded by tilghman (license 14)
+
+2009-05-08 21:01 +0000 [r193391]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c: Set the proper disposition on originated calls.
+	  (closes issue #14167) Reported by: jpt Patches:
+	  call-file-missing-cdr2.diff uploaded by mnicholson (license 96)
+	  Tested by: dlotina, rmartinez, mnicholson
+
+2009-05-08 14:51 +0000 [r193262]  David Vossel <dvossel at digium.com>
+
+	* channels/misdn_config.c: "misdn show config" segfaults asterisk,
+	  if no MSN lists (closes issue #14976) Reported by: alecdavis
+	  Patches: misdn_config.diff.txt uploaded by alecdavis (license
+	  585) Tested by: alecdavis, FabienToune
+
+2009-05-08 14:03 +0000 [r193193]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/logger.conf.sample, main/logger.c: Make absolute paths
+	  for logger channels work properly (Note: This is not a new
+	  feature, it was previously undocumented and broken.) The Asterisk
+	  logger has a feature to support absolute pathnames for logger
+	  channels, but the code implementing the feature was broken. This
+	  has been fixed, and the absolute path feature is now documented
+	  in the sample logger.conf.
+
+2009-05-07 23:41 +0000 [r193119]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c: Fix Background within a Macro for FreePBX. If the
+	  single digit DTMF is an extension in the specified context, then
+	  go there and signal no DTMF. Otherwise, we should exit with that
+	  DTMF. If we're in Macro, we'll exit and seek that DTMF as the
+	  beginning of an extension in the Macro's calling context. If
+	  we're not in Macro, then we'll simply seek that extension in the
+	  calling context. Previously, someone complained about the
+	  behavior as it related to the interior of a Gosub routine, and
+	  the fix (#14011) inadvertently broke FreePBX (#14940). This
+	  change should fix both of these situations, but with the possible
+	  incompatibility that if a single digit extension does not exist
+	  (but a longer extension COULD have matched), it would have
+	  previously gone immediately to the "i" extension, but will now
+	  need to wait for a timeout. (closes issue #14940) Reported by:
+	  p_lindheimer Patches: 20090420__bug14940.diff.txt uploaded by
+	  tilghman (license 14) Tested by: p_lindheimer
+
+2009-05-07 22:17 +0000 [r193050]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: Give a more helpful message when an
+	  incoming call's dialed extension does not match. Added the dialed
+	  extension and context to the chan_misdn messages warning that the
+	  dialed number cannot be matched in the dialplan.
+
+2009-05-07 16:29 +0000 [r192932]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Eliminate repetition of fullcontact during
+	  reconstruction. If the fullcontact field appears in both the
+	  sippeers and the sipregs table, then during reconstruction of the
+	  field, it will otherwise be doubled. (closes issue #14754)
+	  Reported by: Alexei Gradinari Patches:
+	  20090506__bug14754.diff.txt uploaded by tilghman (license 14)
+	  Tested by: lmadsen
+
+2009-05-06 22:15 +0000 [r192858]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_features.c: Make ParkedCall application stop execution of
+	  the dialplan after hang up Just changed park_exec to always
+	  return non-zero. I really wasn't entirely sure at first if this
+	  was a bug. Decided it was since it would be surprising when not
+	  using ParkedCall in the dialplan to hang up and have dialplan
+	  execution continue. (closes issue #14555) Reported by:
+	  francesco_r
+
+2009-05-06 13:30 +0000 [r192633]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Update some old logic to stop both begin and
+	  end DTMF frames from reaching the core if rfc2833 is not enabled.
+	  (closes issue #15036) Reported by: dimas Patches: v1-15036.patch
+	  uploaded by dimas (license 88)
+
+2009-05-05 19:56 +0000 [r192524]  Sean Bright <sean.bright at gmail.com>
+
+	* static-http/astman.js: Fix Javascript error when using astman.js
+	  in Internet Explorer. Internet Explorer (tested with 7.0) does
+	  not like trailing commas on constructs like object initializers,
+	  so get rid of them to avoid some errors. (closes issue #15026)
+	  Reported by: rajnishgiri Patches: bug15026.patch uploaded by
+	  seanbright (license 71) Tested by: seanbright
+
+2009-05-05 18:22 +0000 [r192429-192454]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: Fix an incorrect assumption that certain
+	  values on the channel will always exist when they may not. The
+	  CDR code involved with bridges wrongly assumed that the currently
+	  executing application and data values will always exist. It is
+	  possible for this to be false when call forwarding is involved.
+	  (closes issue #14984) Reported by: gincantalupo
+
+	* apps/app_followme.c: Fix a bug where the followme application
+	  would continue trying numbers after the caller hung up. (closes
+	  issue #13624) Reported by: sgenyuk
+
+2009-05-04 22:37 +0000 [r192213]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: global mohinterpret setting is ignored
+	  mohinterpret and mohsuggest global variables were not copied over
+	  during build_users and build_peers. (closes issue #14728)
+	  Reported by: dimas Patches: v1-14728.patch uploaded by dimas
+	  (license 88) Tested by: dimas, dvossel
+
+2009-05-02 18:48 +0000 [r191628-191778]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Fix a bug which resulted from the Hebrew
+	  voicemail commit. This fixes a case where a certain message could
+	  get played twice. (closes issue #13155) Reported by:
+	  greenfieldtech Patches: app_voicemail.c.multi-lang-patch uploaded
+	  by greenfieldtech (license 369) Tested by: greenfieldtech
+
+	* apps/app_chanspy.c: Kevin has informed me that thi sort of thing
+	  is not necessary.
+
+	* apps/app_chanspy.c: Move static buffers to outside for loops in
+	  app_chanspy. Similar to seanbright's commit 191422, this moves
+	  some static buffers to be defined outside of for loops since it
+	  is undefined if memory will be re-used or if the stack will grow
+	  with each iteration of the loop.
+
+2009-05-01 20:00 +0000 [r191559]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: SIP Response 410 maps to cause code 22 (or
+	  23), not 1. (closes issue #14993) Reported by: BigJimmy Patches:
+	  causepatch uploaded by BigJimmy (license 371)
+
+2009-05-01 17:40 +0000 [r191488]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: Fix DTMF not being sent to other side after a
+	  partial feature match This fixes a regression from commit 176701.
+	  The issue was that ast_generic_bridge never exited after the
+	  feature digit timeout had elapsed, which prevented the queued
+	  DTMF from being sent to the other side. This issue was reported
+	  to me directly.
+
+2009-05-01 15:42 +0000 [r191422]  Sean Bright <sean.bright at gmail.com>
+
+	* apps/app_queue.c: Move the defintion of the a couple arrays out
+	  of loops. According to Kevin, it is unspecified as to whether a
+	  variable defined inside a block is allocated once by the compiler
+	  or for each pass through the block (loops being the only
+	  interesting case), so just define these before we get into our
+	  loop to be sure.
+
+2009-04-29 23:10 +0000 [r191220]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/h323/ast_h323.cxx, channels/chan_h323.c: Allow H.323 to
+	  compile with FDLEAK checking enabled.
+
+2009-04-29 18:07 +0000 [r191096]  David Brooks <dbrooks at digium.com>
+
+	* pbx/pbx_config.c: Patch to fix tab-completion crash on "remove
+	  extension" This patch simply removes some old code back before
+	  Asterisk used editline. This fixes the crash that occurred when
+	  tab-completing "remove extension". (closes issue #14689) Reported
+	  by: isaacgal
+
+2009-04-29 15:23 +0000 [r191041]  Sean Bright <sean.bright at gmail.com>
+
+	* apps/app_queue.c: Fix a crash in app_queue with very long member
+	  lists. A user reported via #asterisk that with very long lists of
+	  members, a crash occurs in ast_strdupa, so just use a single
+	  buffer and ast_copy_string instead of stack allocating copys of
+	  each interface name.
+
+2009-04-27 19:29 +0000 [r190721]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in: Fix 'inconsistent
+	  line endings' when autoconf 2.63 is used Attempt to make
+	  configure script regeneration 'safe' using autoconf 2.63, which
+	  embeds a bare CR into the script, thus making Subversion complain
+	  about inconsistent line endings This commit changes the MIME type
+	  of the configure script to be 'binary' thus making Subversion no
+	  longer inspect line endings, and as a bonus 'svn diff' will no
+	  longer try to generate diff output for it, which is not generally
+	  useful anyway.
+
+2009-04-27 19:03 +0000 [r190661-190662]  Russell Bryant <russell at digium.com>
+
+	* res/res_smdi.c: Fix a typo from 190661.
+
+	* res/res_smdi.c: Resolve a crash in res_smdi when used with
+	  chan_dahdi. When chan_dahdi goes to get an SMDI message, it
+	  provides no search criteria. It just grabs the next message that
+	  arrives. This code was written with the SMDI dialplan functions
+	  in mind, since that is now the preferred method of using SMDI.
+	  However, this broke support of it being used from chan_dahdi.
+	  (closes AST-212)
+
+2009-04-23 21:07 +0000 [r190356]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Remove a bogus ast_channel_unlock().
+
+2009-04-23 19:13 +0000 [r190286]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_local.c: Fix a bug in chan_local glare hangup
+	  detection. If both sides of a Local channel were hung up at
+	  around the same time it was possible for one thread to destroy
+	  the local private structure and have the other thread immediately
+	  try to remove the already freed structure from the local channel
+	  list.
+
+2009-04-23 10:07 +0000 [r190187]  Olle Johansson <oej at edvina.net>
+
+	* include/asterisk/lock.h: unistd.h is required for usleep() on
+	  Darwin. It will not hurt to include it always on other platforms
+	  either.
+
+2009-04-22 21:35 +0000 [r190092]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  include/asterisk/lock.h: Detect availability of
+	  pthread_rwlock_timedwrlock() before using it. (closes issue
+	  #14930) Reported by: tilghman Patches:
+	  20090420__bug14930.diff.txt uploaded by tilghman (license 14)
+	  Tested by: mvanbaak, tilghman
+
+2009-04-22 19:20 +0000 [r189991]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/h323/ast_h323.cxx, channels/chan_h323.c,
+	  channels/h323/chan_h323.h: Make chan_h323 respect packetization
+	  settings Previously, packetization settings were ignored and now
+	  they are not. A new config option 'autoframing' has been added to
+	  mirror the way chan_sip handles it. Turning on the autoframing
+	  option (available both as a global option or per peer) overrides
+	  the local settings with the remote packetization settings.
+	  Testing was performed with varying packetization levels with the
+	  following codecs: ulaw, alaw, gsm, and g729. (closes issue
+	  #12415) Reported by: pj Patches:
+	  2009012200_h323packetization.diff.txt uploaded by mvanbaak
+	  (license 7), modified by me
+
+2009-04-22 14:29 +0000 [r189849]  Michiel van Baak <michiel at vanbaak.info>
+
+	* contrib/scripts/get_ilbc_source.sh: replace sed with tr to remove
+	  \r from downloaded file On some systems, sed does not recognize
+	  \r in the pattern the way it was used here. Use tr instead
+	  because this works the same across systems. (closes issue #14936)
+	  Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded
+	  by mvanbaak (license 7) Tested by: leobrown, mvanbaak
+
+2009-04-21 15:52 +0000 [r189601-189664]  Doug Bailey <dbailey at digium.com>
+
+	* utils/muted.c: Remove daemon call on systems that do not support
+	  forking.
+
+	* main/config.c, configure, include/asterisk/autoconfig.h.in,
+	  include/asterisk/compat.h, configure.ac: Add check in configure
+	  script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This
+	  allows config.c to compile when linked against uclibc that does
+	  not support these parameters
+
+2009-04-20 22:02 +0000 [r189537]  Tilghman Lesher <tlesher at digium.com>
+
+	* funcs/func_odbc.c, funcs/func_strings.c: Add a workaround for
+	  func_odbc/ARRAY() for problems that occur with certain special
+	  characters. In certain cases, due to the way Set() works in 1.4,
+	  values may not get set properly. This is a workaround for 1.4
+	  only that corrects for these issues, without making func_odbc
+	  more difficult to use properly. (closes issue #14614) Reported
+	  by: wdoekes Patches: 20090309__bug14614__2.diff.txt uploaded by
+	  tilghman (license 14)
+	  double_set_unescape_workaround_for_func_odbc.osso-and-tilghman-1.diff
+	  uploaded by wdoekes (license 717) Tested by: wdoekes, tilghman
+
+2009-04-20 21:10 +0000 [r189463-189465]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_dial.c: Update CDR appropriately when
+	  AST_CAUSE_NO_ANSWER is set
+
+	* apps/app_dial.c: Don't treat a NOANSWER like a CHANUNAVAIL
+
+2009-04-20 20:58 +0000 [r189462]  Sean Bright <sean.bright at gmail.com>
+
+	* pbx/ael/ael.tab.c, pbx/ael/ael.y: Properly handle @s within hints
+	  in AEL. AEL was not handling the case of a device hint containing
+	  an @ symbol, which caused parking hints (e.g.
+	  hint(park:exten at context)) to error out the parser. This patch
+	  makes AEL treat the @ the same way it treats colon and ampersand
+	  now, meaning the characters are included in verbatim. (closes
+	  issue #14941) Reported by: bpgoldsb Patches: bug14941.patch
+	  uploaded by seanbright (license 71) Tested by: bpgoldsb
+
+2009-04-20 19:10 +0000 [r189391]  Doug Bailey <dbailey at digium.com>
+
+	* main/manager.c, main/db1-ast/recno/rec_open.c,
+	  channels/chan_iax2.c: Clean up problem with manager
+	  implementation of mmap where it was not testing against
+	  MAP_FAILED response. Got rid of shadowed variable used in
+	  processign the mmap results. Change test of mmap results to
+	  compare against MAP_FAILED
+
+2009-04-20 14:04 +0000 [r189277]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c: Move the check for chan->fdno == -1 to after the
+	  zombie/hangup check. Many users were finding that their hung up
+	  channels were staying up and causing 100% CPU usage. (issue
+	  #14723) Reported by: seadweller Patches: 14723_1-4-tip.patch
+	  uploaded by mmichelson (license 60) Tested by: falves11, bamby
+
+2009-04-18 01:27 +0000 [r189203]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_agent.c: Fixed autologoff in agents.conf not
+	  working when agent logs in via AgentLogin app An agent logs in by
+	  calling an extension that calls the AgentLogin app. In
+	  agents.conf ackcall=always is set, so when they get a call they
+	  have the choice to either acknowledge it or ignore it.
+	  autologoff=10 is set as well, so if the agent ignores the call
+	  over 10sec one may assume that the agent should be logged out
+	  (and in this case hungup on as well), but this was not happening.
+	  (closes issue #14091) Reported by: evandro Patches:
+	  autologoff.diff uploaded by dvossel (license 671) Review:
+	  http://reviewboard.digium.com/r/225/
+
+2009-04-17 21:27 +0000 [r189134]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c: Modifed/added some debug messages.
+	  JIRA ABE-1835
+
+2009-04-17 15:43 +0000 [r189009]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c: Make Busy() application set the CDR disposition to
+	  BUSY. (closes issue #14306) Reported by: cristiandimache
+
+2009-04-17 14:41 +0000 [r188937-188946]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Fix a bug where a value used to create the
+	  channel name was bogus. This commit fixes the scenario where an
+	  incoming call is authenticated using a peer entry. Previously the
+	  channel name was created using either the username setting from
+	  the sip.conf entry or the IP address that the call came from. Now
+	  the channel name will be created using the peer name itself. This
+	  commit will not change the way the channel name is generated for
+	  users or friends. (closes issue #14256) Reported by: Nick_Lewis
+	  Patches: chan_sip.c-chname.patch uploaded by Nick (license 657)
+	  Tested by: Nick_Lewis, file
+
+	* channels/chan_dahdi.c: Fix a situation where the DAHDI channel
+	  private structure lock was not unlocked when it should have been.
+	  (issue AST-210)
+
+2009-04-16 21:41 +0000 [r188835]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Only update realtime, if global option
+	  rtupdate != false (closes issue #14885) Reported by: deepesh
+	  Patches: 20090413__bug14885.diff.txt uploaded by tilghman
+	  (license 14) Tested by: deepesh
+
+2009-04-16 21:37 +0000 [r188833]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_misdn.c: Only disable mISDN DSP if Asterisk DSP is
+	  enabled. Leave jitter setting alone. JIRA ABE-1835
+
+2009-04-16 21:02 +0000 [r188773]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Umask should not be exported into global
+	  namespace. (closes issue #14912) Reported by: jcapp
+
+2009-04-15 22:08 +0000 [r188646]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c: National prefix inserted even when caller
+	  ID not available When the caller ID is restricted, the expected
+	  behavior is for the caller id to be blank. In chan_dahdi, the
+	  national prefix is placed onto the callers number even if its
+	  restricted (empty) causing the caller id to be the national
+	  prefix rather than blank. (closes issue #13207) Reported by:

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