[asterisk-commits] bebuild: tag 10.4.0-digiumphones-rc1 r365260 - /tags/10.4.0-digiumphones-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu May 3 14:42:16 CDT 2012


Author: bebuild
Date: Thu May  3 14:42:10 2012
New Revision: 365260

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=365260
Log:
Importing files for 10.4.0-digiumphones-rc1 release.

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    tags/10.4.0-digiumphones-rc1/.version   (with props)
    tags/10.4.0-digiumphones-rc1/ChangeLog   (with props)

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--- tags/10.4.0-digiumphones-rc1/ChangeLog (added)
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+2012-05-03  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.4.0-digiumphones-rc1 Released.
+
+2012-05-03 19:14 +0000 [r365214]  Jason Parker <jparker at digium.com>
+
+	* main/app.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
+	  channels/chan_iax2.c, res/res_fax_spandsp.c,
+	  apps/app_milliwatt.c, main/channel.c, pbx/pbx_loopback.c,
+	  addons/chan_ooh323.c, main/manager.c, /, channels/chan_sip.c,
+	  channels/chan_skinny.c: Multiple revisions
+	  361143,363103-363104,363107,363156,364707 ........ r361143 |
+	  jrose | 2012-04-04 11:38:12 -0500 (Wed, 04 Apr 2012) | 12 lines
+	  Replace GNU old-style field designator extensions to fix clang
+	  warnings (issue ASTERISK-19540) Reported by: Makoto Dei Patches:
+	  clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
+	  ........ Also add from the patch the portion in res_fax_spandsp
+	  that didn't apply to 1.8 Merged revisions 361142 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
+	  ASTERISK-19540) ........ r363103 | mjordan | 2012-04-23 08:40:23
+	  -0500 (Mon, 23 Apr 2012) | 19 lines AST-2012-005: Fix remotely
+	  exploitable heap overflow in keypad button handling When handling
+	  a keypad button message event, the received digit is placed into
+	  a fixed length buffer that acts as a queue. When a new message
+	  event is received, the length of that buffer is not checked
+	  before placing the new digit on the end of the queue. The
+	  situation exists where sufficient keypad button message events
+	  would occur that would cause the buffer to be overrun. This patch
+	  explicitly checks that there is sufficient room in the buffer
+	  before appending a new digit. (closes issue ASTERISK-19592)
+	  Reported by: Russell Bryant ........ Merged revisions 363100 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+	  Merged revisions 363102 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r363104 | mjordan | 2012-04-23 08:48:48 -0500 (Mon, 23 Apr 2012)
+	  | 10 lines Reference skinny_subchannel object instead of
+	  skinny_device for r363103 The check-in to resolve ASTERISK-19592
+	  (r363103) failed to switch to the skinny_subchannel object
+	  instead of the skinny_device when attempting to reference the
+	  buffer for the keypad digits. This patch fixes that. (issue
+	  ASTERISK-19592) Reported by: Russell Bryant ........ r363107 |
+	  mjordan | 2012-04-23 09:07:29 -0500 (Mon, 23 Apr 2012) | 19 lines
+	  AST-2012-006: Fix crash in UPDATE handling when no channel owner
+	  exists If Asterisk receives a SIP UPDATE request after a call has
+	  been terminated and the channel has been destroyed but before the
+	  SIP dialog has been destroyed, a condition exists where a
+	  connected line update would be attempted on a non-existing
+	  channel. This would cause Asterisk to crash. The patch resolves
+	  this by first ensuring that the SIP dialog has an owning channel
+	  before attempting a connected line update. If an UPDATE request
+	  is received and no channel is associated with the dialog, a 481
+	  response is sent. (closes issue ASTERISK-19770) Reported by:
+	  Thomas Arimont Tested by: Matt Jordan Patches:
+	  ASTERISK-19278-2012-04-16.diff uploaded by Matt Jordan (license
+	  6283) ........ Merged revisions 363106 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r363156 | jrose | 2012-04-23 09:39:48 -0500 (Mon, 23 Apr 2012) |
+	  23 lines AST-2012-004: Fix an error that allows AMI users to run
+	  shell commands sans authorization. As detailed in the advisory,
+	  AMI users without write authorization for SYSTEM class AMI
+	  actions were able to run system commands by going through other
+	  AMI commands which did not require that authorization.
+	  Specifically, GetVar and Status allowed users to do this by
+	  setting their variable/s options to the SHELL or EVAL functions.
+	  Also, within 1.8, 10, and trunk there was a similar flaw with the
+	  Originate action that allowed users with originate permission to
+	  run MixMonitor and supply a shell command in the Data argument.
+	  That flaw is fixed in those versions of this patch. (closes issue
+	  ASTERISK-17465) Reported By: David Woolley Patches:
+	  162_ami_readfunc_security_r2.diff uploaded by jrose (license
+	  6182) 18_ami_readfunc_security_r2.diff uploaded by jrose (license
+	  6182) 10_ami_readfunc_security_r2.diff uploaded by jrose (license
+	  6182) ........ Merged revisions 363117 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
+	  Merged revisions 363141 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
+	  r364707 | mmichelson | 2012-04-30 14:42:35 -0500 (Mon, 30 Apr
+	  2012) | 17 lines Revert improved identities sent in dialog-info
+	  NOTIFY requests in r360862 Revision 360862 was intended to
+	  improve identities sent in dialog-info NOTIFY requests. Some
+	  users reported that hint became broken once this was done. It's
+	  not clear exactly what part of the patch has caused this
+	  regression, but broken hints are bad. For now, this revision is
+	  being reverted so that the next releases of Asterisk do not have
+	  bad behavior in them. The original reported issue will have to be
+	  fixed differently in the next version of Asterisk. (issue
+	  ASTERISK-16735) ........ Merged revisions 364706 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 361143,363103-363104,363107,363156,364707 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+2012-05-02 15:57 +0000 [r364766-365009]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c: Save the address on which a MESSAGE was
+	  received, so it can be used in MESSAGE() This is useful in cases
+	  where chan_sip may be listening on multiple addresses. (issue
+	  DPMA-127)
+
+	* main/manager.c, /: Prevent a potential crash when using manager
+	  hooks. Found by me while poking at DPMA-127. ........ Merged
+	  revisions 364841 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 364842 from
+	  http://svn.asterisk.org/svn/asterisk/branches/10
+
+	* tests/test_voicemail_api.c, include/asterisk/app_voicemail.h,
+	  apps/app_voicemail.c, /: Remove folder_dir from voicemail
+	  snapshots API. It was both unused (except in tests, where it was
+	  fudged) and unnecessary. (closes issue AST-842) ........ Merged
+	  revisions 364761 from
+	  http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
+
+2012-04-19 20:35 +0000 [r362674]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: The 10-digiumphones version of revision
+	  362673 to 1.8-digiumphones
+
+2012-04-13 17:48 +0000 [r362134]  Jason Parker <jparker at digium.com>
+
+	* /: init merge prop
+
+2012-04-13 17:29 +0000 [r362133]  Matthew Jordan <mjordan at digium.com>
+
+	* /: Add branch-10-merged property
+
+2012-04-09 21:32 +0000 [r361748]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.exports.in, tests/test_voicemail_api.c
+	  (added), include/asterisk/app_voicemail.h, apps/app_voicemail.c,
+	  main/manager.c: Fix bugs in voicemail APIs and add unit tests.
+
+2012-04-06 19:16 +0000 [r361503]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/message.c: Update Func MESSAGE() and AMI MessageSend
+	  documentation. * Document MESSAGE(custom_data) * Update AMI
+	  MessageSend documentation * Eliminate a shadowed variable name in
+	  msg_func_write() for custom_data.
+
+2012-04-05 17:48 +0000 [r361140-361293]  Mark Michelson <mmichelson at digium.com>
+
+	* tests/test_config.c, funcs/func_presencestate.c: Add unit tests
+	  for func_presencestate and config hooks. These tests were
+	  originally written for the trunk merge of Digium phone support to
+	  Asterisk, but this branch can benefit from these tests too.
+
+	* main/pbx.c: Fix a compiler error in pbx.c This places a function
+	  declaration before its first invocation.
+
+	* main/pbx.c: Only add the device portion of a hint when adding a
+	  hintdevice.
+
+	* main/file.c, apps/app_queue.c, apps/app_mixmonitor.c,
+	  main/message.c, apps/app_voicemail.c, channels/chan_sip.c,
+	  channels/chan_skinny.c: Fix compiler errors. Next up is to do
+	  some testing to be sure things are sane.
+
+	* channels/sip/include/sip.h, channels/chan_sip.c,
+	  channels/chan_skinny.c: Changes to the channels directory for
+	  Digium phone support. Next step: Get this poopoo compiling!
+
+	* tests/test_config.c (added), tests/test_custom_control.c (added):
+	  Add tests for Digium phone support.
+
+	* apps/app_voicemail.exports.in, apps/app_queue.c,
+	  apps/app_mixmonitor.c, apps/app_voicemail.c: apps changes
+	  necessary for Digium phone support.
+
+	* funcs/func_frame_trace.c, funcs/func_presencestate.c (added): Add
+	  Digium phone changes for the funcs directory.
+
+	* include/asterisk/custom_control_frame.h (added),
+	  include/asterisk/message.h, include/asterisk/config.h,
+	  include/asterisk/presencestate.h (added),
+	  include/asterisk/app_voicemail.h (added), include/asterisk/pbx.h,
+	  include/asterisk/frame.h, include/asterisk/manager.h,
+	  include/asterisk/file.h, include/asterisk/app.h,
+	  include/asterisk/event_defs.h: Add Digium phone changes for the
+	  include/asterisk directory
+
+	* main/message.c: Add phone changes for message.c
+
+	* main/features.c, main/file.c, main/app.c, main/event.c,
+	  main/custom_control_frame.c (added), main/asterisk.c,
+	  main/config.c, main/presencestate.c (added), main/channel.c,
+	  main/pbx.c, main/manager.c: Update everything in the main
+	  directory for Digium phone additions.
+
+	* configs/manager.conf.sample: Update sample config files. This
+	  just involved updating manager.conf.sample to have the new
+	  "message" write permission.
+
+	* / (added): Create a Digium phones branch of Asterisk 10.
+
+2012-03-29  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.3.0 Released.
+
+2012-03-26  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.3.0-rc3 Released.
+
+	* AST-2012-003
+
+	* AST-2012-002
+
+	* /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock
+	  regression by allowing AMI action callback to be reentrant
+
+	  Fix AMI module reload deadlock from ASTERISK-18479 when it tired to
+	  fix the race between calling an AMI action callback and
+	  unregistering that action.  Refixes ASTERISK-13874 broken by
+	  ASTERISK-17785 change.
+
+	  Locking the ao2 object guaranteed that there were no active
+	  callbacks that mattered when ast_manager_unregister() was called.
+	  Unfortunately, this causes the deadlock situation.  The path stops
+	  locking the ao2 object to allow multiple threads to invoke the
+	  callback re-entrantly.  There is no way to guarantee a module unload
+	  will not crash because of an active callback.  The code attempts to
+	  minimize the chance with the registered flag and the maximum 5
+	  second delay before ast_manager_unregister() returns.
+
+	  The trunk version of the patch changes the API to fix the race
+	  condition correctly to prevent the module code from unloading from
+	  memory while an action callback is active.
+
+	  * Don't hold the lock while calling the AMI action callback.
+
+	  (closes issue ASTERISK-19487)
+	  Reported by: Philippe Lindheimer
+
+	  Review: https://reviewboard.asterisk.org/r/1818/
+
+2012-03-06  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.3.0-rc2 Released.
+
+	* main/acl.c: Prevent outbound SIP NOTIFY packets from displaying
+	  a port of 0.
+
+	  In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which
+	  changed the behavior of ast_find_ourip such that port number was
+	  wiped out.  This caused the port in internip (which is used for
+	  Contact and Call-ID on NOTIFYs) to be 0.  This change causes
+	  ast_find_ourip to be port-preserving again.
+
+2012-01-30 22:16 +0000 [r353369-353321]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/sip/include/dialog.h, /, channels/chan_sip.c,
+	  channels/sip/include/sip.h: Merged revisions 353320 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
+	  ........ r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31
+	  Jan 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number
+	  value MUST be expressible as a 32-bit unsigned integer * fix: use
+	  %u instead of %d when dealing with CSeq numbers - to remove
+	  possibility of -ve numbers. * fix: change all uses of seqno and
+	  friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
+	  Summary of CSeq numbers. An initial CSeq number must be less than
+	  2^31 A CSeq number can increase in value up to 2^32-1 An
+	  incrementing CSeq number must not wrap around to 0. Tested with
+	  Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
+	  Tested by: alecdavis Review:
+	  https://reviewboard.asterisk.org/r/1699/ ........
+
+	* /, channels/chan_sip.c: Merged revisions 353368 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan
+	  2012) | 2 lines prevent debug messsges displaying -ve Cseq
+	  numbers. Missed in R353320 ........
+
+2012-01-30 23:28 +0000 [r353397]  Terry Wilson <twilson at digium.com>
+
+	* main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
+	  Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
+	  currently takes a pointer to an ast_sockaddr and updates it
+	  anytime an address resolves to something different. There are a
+	  couple of issues with this. First, the ast_sockaddr is usually
+	  the address of an ast_sockaddr inside a refcounted struct and we
+	  never bump the refcount of those structs when using dnsmgr. This
+	  makes it possible that a refresh could happen after the
+	  destructor for that object is called (despite ast_dnsmgr_release
+	  being called in that destructor). Second, the module using dnsmgr
+	  cannot be aware of an address changing without polling for it in
+	  the code. If an action needs to be taken on address update (like
+	  re-linking a SIP peer in the peers_by_ip table), then polling for
+	  this change negates many of the benefits of having dnsmgr in the
+	  first place. This patch adds a function to the dnsmgr API that
+	  calls an update callback instead of blindly updating the address
+	  itself. It also moves calls to ast_dnsmgr_release outside of the
+	  destructor functions and into cleanup functions that are called
+	  when we no longer need the objects and increments the refcount of
+	  the objects using dnsmgr since those objects are stored on the
+	  ast_dnsmgr_entry struct. A helper function for returning the
+	  proper default SIP port (non-tls vs tls) is also added and used.
+	  This patch also incorporates changes from a patch posted by Timo
+	  Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
+	  ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
+	  ........ Merged revisions 353371 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-01-31 17:21 +0000 [r353463]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/manager.c, /, include/asterisk/channel.h: Fix memory leak in
+	  error paths for action_originate(). * Fix memory leak of vars in
+	  error paths for action_originate(). * Moved struct
+	  fast_originate_helper tech and data members to stringfields. *
+	  Simplified ActionID header handling for fast_originate(). * Added
+	  doxygen note to ast_request() and ast_call() and the associated
+	  channel callbacks that the data/addr parameters should be treated
+	  as const char *. Review: https://reviewboard.asterisk.org/r/1690/
+	  ........ Merged revisions 353454 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 00:00 +0000 [r353503]  Terry Wilson <twilson at digium.com>
+
+	* res/res_calendar.c, /: Allow res_calendar to be unloaded The
+	  calendaring tech modules depend on res_calendar and initially
+	  res_calendar just bumped the use count so that it couldn't be
+	  unloaded. res_calendar can potentially create many threads and
+	  I've seen issues where the Asterisk shutdown has failed where it
+	  looked like these threads could be the culprit. This patch adds
+	  unload support for res_calendar. Unloading res_calendar will also
+	  unload the dependant tech modules as well. (closes issue
+	  ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
+	  ........ Merged revisions 353502 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:05 +0000 [r353551]  Matthew Jordan <mjordan at digium.com>
+
+	* /, contrib/init.d/etc_default_asterisk: Added clarification for
+	  the VERBOSITY setting to etc_default_asterisk Clarified that
+	  using the VERBOSITY setting in etc_default_asterisk is the same
+	  as using the -v command line switch, which causes Asterisk to
+	  launch in console mode. (closes issue ASTERISK-17030) Reported
+	  by: Jonas ........ Merged revisions 353550 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 15:51 +0000 [r353599]  Sean Bright <sean at malleable.com>
+
+	* /, include/asterisk/audiohook.h: Resolve an overlap in the
+	  ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
+	  AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
+	  unintended side effects. This patch moves
+	  AST_AUDIOHOOK_TRIGGER_WRITE, and updates
+	  AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
+	  This will affect existing modules that use these flags, so be
+	  sure to recompile as necessary. (closes issue ASTERISK-19246)
+	  Reported by: feyfre ........ Merged revisions 353598 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-01 21:16 +0000 [r353771-353721]  Jonathan Rose <jrose at digium.com>
+
+	* /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers
+	  for various functions in chan_sip There are a number of cleaner
+	  looking wrappers for ast_sockaddr_stringify_fmt available which
+	  are slightly more readable than using a direct call to
+	  ast_sockaddr_stringify_fmt. This patch switches a number of those
+	  calls in chan_sip to use those wrappers and is generally
+	  harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
+	  Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
+	  Michael L. Young (license 5026) ........ Merged revisions 353720
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Fix sip show peers port output, align
+	  columns, and fix ami port output. A previous patch I committed
+	  from ASTERISK-16930 unexpectedly changed some output for the AMI
+	  action "sippeers" which this patch changes back. Also, this
+	  aligns the output for the cli command "sip show peers" and fixes
+	  another issue that patch introduced by using
+	  ast_sockaddr_stringify calls multiple times without immediately
+	  using the pointer. I also went ahead and did a little janitorial
+	  work to clean up whitespace in _sip_show_peers. (issue
+	  ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
+	  Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
+	  Walter Doekes (license 5674) ........ Merged revisions 353769
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 18:48 +0000 [r353820]  Mark Michelson <mmichelson at digium.com>
+
+	* configs/http.conf.sample, main/manager.c, /, main/http.c,
+	  configs/manager.conf.sample, include/asterisk/manager.h: Fix TLS
+	  port binding behavior as well as reload behavior: * Removes
+	  references to tlsbindport from http.conf.sample and
+	  manager.conf.sample * Properly bind to port specified in
+	  tlsbindaddr, using the default port if specified. * On a reload,
+	  properly close socket if the service has been disabled. A note
+	  has been added to UPGRADE.txt to indicate how ports must be set
+	  for TLS. (closes issue ASTERISK-16959) reported by Olaf
+	  Holthausen (closes issue ASTERISK-19201) reported by Chris
+	  Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas
+	  Review: https://reviewboard.asterisk.org/r/1709 ........ Merged
+	  revisions 353770 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 20:11 +0000 [r353868]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
+	  Restore the 'w' modifier support for ISDN spans.
+	  Dial(DAHDI/g0/1234w888) This feature also causes the sending
+	  complete ie to be sent for switch types that do not automatically
+	  send the ie. (EuroISDN/ETSI) The main difference between dialing
+	  Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
+	  sending of the sending complete ie. (closes issue ASTERISK-19176)
+	  Reported by: rmudgett Tested by: rmudgett ........ Merged
+	  revisions 353867 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-02 22:27 +0000 [r353916]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
+	  cause an infinite loop After R340970 Asterisk was still polling
+	  the RTCP file descriptor after RTCP is shut down and removed. If
+	  the descriptor happened to have data ready when the removal
+	  occured then Asterisk would go into an infinite loop trying to
+	  read data that it can never actually access. This change disables
+	  the audio RTCP file descriptor for the duration of the T.38
+	  transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
+	  Vrban ........ Merged revisions 353915 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-03 16:22 +0000 [r354000-353962]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_fax.c: Fixes a segfault occuring when performing attended
+	  transfer with FAXOPT(gateway)=yes (closes issue ASTERISK-19184)
+	  Reported by: Alexandr
+
+	* /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
+	  due to r335976 Bad locking order was added to chan_agent to
+	  prevent segfaults from having no locking in a patch by irroot.
+	  This patch addresses the bad locking order by releasing locks
+	  before getting the right locking order to stop deadlocks from
+	  occuring when doing multiple interactions with agents. (closes
+	  issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
+	  https://reviewboard.asterisk.org/r/1708/ ........ Merged
+	  revisions 353999 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-06 17:31 +0000 [r354217-354119]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/features.c: Add missing headers to AMI UnParkedCall event
+	  to uniquely identify the call. The AMI UnParkedCall event was
+	  missing the Parkinglot and Uniqueid headers that the AMI
+	  ParkedCall event contains. (closes issue ASTERISK-19240) Reported
+	  by: Michael Yara ........ Merged revisions 354116 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
+	  extension" command. * Documented dialplan add extension
+	  <exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
+	  of command without the app-data value. There are many
+	  applications that do no need any parameters so it is silly to
+	  require that field for all commands. * Fixed a couple
+	  ast_malloc/ast_free mismatches with ast_add_extension2() calls.
+	  (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
+	  by: rmudgett ........ Merged revisions 354216 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 15:19 +0000 [r354270]  Jonathan Rose <jrose at digium.com>
+
+	* /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
+	  for cdr_pgsql. Prior to this patch, attempts to reload
+	  cdr_pgsql.so would cause the column list to keep its current data
+	  and then add a second copy during the reload. This would cause
+	  attempts to log the CDR to the database to fail. This patch also
+	  cleans up some unnecessary null checks for ast_free and deals
+	  with a few potential locking problems. (closes issue
+	  ASTERISK-19216) Reported by: Jacek Konieczny Review:
+	  https://reviewboard.asterisk.org/r/1711/ ........ Merged
+	  revisions 354263 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-07 21:17 +0000 [r354349]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
+	  Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
+	  instead of "" 2. Don't set ipaddr or port to the string "(null)"
+	  when they are empty 3. Add missing required fields, set default
+	  for lastms to 0, and modify the length of the ipaddr field to 45
+	  in the Postgresql realtime.sql file. (closes issue
+	  ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
+	  ........ Merged revisions 354348 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:25 +0000 [r354493]  Russell Bryant <russell at russellbryant.com>
+
+	* main/channel.c, /: Remove some unnecessary locking from
+	  ast_hangup(). This patch removes some unnecessary locking of the
+	  channels container in ast_hangup(). The reason this came up is
+	  that this lock can very quickly block the entire system. If any
+	  of the channel cleanup code decides to block, it causes a problem
+	  for the whole system. For example, when audiohooks get destroyed,
+	  if that blocks for a while waiting on the mixmonitor thread to
+	  exit because it's busy blocking on some I/O, it causes a problem
+	  for many other threads in the meantime. Review:
+	  https://reviewboard.asterisk.org/r/1712/ ........ Merged
+	  revisions 354492 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 02:54 +0000 [r354496]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_parkandannounce.c, /: Fix crash in ParkAndAnnounce.
+	  Well, thats embarrasing. I forgot to initialize the caller_id
+	  storage. (closes issue ASTERISK-19311) Reported by: tootai Tested
+	  by: rmudgett ........ Merged revisions 354495 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 16:35 +0000 [r354543]  Matthew Jordan <mjordan at digium.com>
+
+	* /, channels/chan_sip.c: Fix SIP INFO DTMF handling for
+	  non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as
+	  changed to account for both lowercase alphatbetic DTMF events, as
+	  well as uppercase alphabetic DTMF events. When this occurred, the
+	  comparison of the character buffer containing the event code was
+	  changed such that the buffer was first compared again '0' and '9'
+	  to determine if it was numeric. Unfortunately, since the first
+	  character in the buffer will typically be '1' in the case of
+	  non-numeric event codes (10-16), this caused those codes to be
+	  converted to a DTMF event of '1'. This patch fixes that, and
+	  cleans up handling of both application/dtmf-relay and
+	  application/dtmf content types. Review:
+	  https://reviewboard.asterisk.org/r/1722/ (closes issue
+	  ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........
+	  Merged revisions 354542 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:04 +0000 [r354546]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_fax.c: Adding reload support to res_fax.so (closes
+	  issue ASTERISK-16712) reported by Frank DiGennaro Review:
+	  https://reviewboard.asterisk.org/r/1713 ........ Merged revisions
+	  354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 17:08 +0000 [r354548]  Matthew Jordan <mjordan at digium.com>
+
+	* /, channels/chan_sip.c: Clean-up of minor formatting issues in
+	  r354542/3/4 rmudgett pointed out some formatting issues in the
+	  check-in for ASTERISK-19290. This cleans those up. Review:
+	  https://reviewboards.asterisk.org/r/1722/ ........ Merged
+	  revisions 354547 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 19:54 +0000 [r354703-354656]  Kinsey Moore <kmoore at digium.com>
+
+	* /, main/config.c: Make the config parser remove escaping
+	  backslashes The config parser in Asterisk does not currently
+	  remove a backslash that is used to escape a semicolon which would
+	  otherwise be interpreted as the start of a comment. The change
+	  here causes that backslash to be removed, but does not create a
+	  real escape system in the config parser. The biggest complication
+	  with a real escape system would be breaking existing configs
+	  everywhere (parsing \\ as \ and breaking on escaped non-semicolon
+	  characters) even though it would be the "right" way to do things.
+	  (closes issue ASTERISK-17121) Review:
+	  https://reviewboard.asterisk.org/r/1724/ ........ Merged
+	  revisions 354655 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c: Fix parsing of SIP headers where compact
+	  and non-compact headers are mixed Change parsing of SIP headers
+	  so that compactness of the header no longer influences which
+	  header will be chosen. Previously, a non-compact header would be
+	  chosen instead of a preceeding compact-form header. (closes issue
+	  ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
+	  ........ Merged revisions 354702 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-09 22:03 +0000 [r354750]  Terry Wilson <twilson at digium.com>
+
+	* /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
+	  is torn down CDRs cannot be modified after a bridge is torn down,
+	  (e.g. after Dial() returns) even though the CDR() function may be
+	  called. Since modifying the CDR code to change this behavior
+	  could very easily break all kinds of things, this patch just
+	  documents this limitation. (closes issues ASTERISK-16923) Review:
+	  https://reviewboard.asterisk.org/r/1720/ ........ Merged
+	  revisions 354749 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 18:05 +0000 [r354836]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
+	  to the same exten and context. The astman_get_header() never
+	  returns NULL so the check by the code for NULL would never fail.
+	  (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
+	  0018325.patch (license #6116) patch uploaded by Nuno Borges
+	  (modified) ........ Merged revisions 354835 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-10 22:00 +0000 [r354890]  Jason Parker <jparker at digium.com>
+
+	* apps/app_voicemail.c, /: Fix a voicemail memory leak with
+	  heard/deleted messages. open_mailbox() was changed quite a long
+	  time ago to allocate this memory. close_mailbox() should have
+	  been changed to be responsible for freeing it. ........ Merged
+	  revisions 354889 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 16:41 +0000 [r354938]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_confbridge.c: Don't try to play sound files that do not
+	  exist. (closes issue ASTERISK-19188) Reported by: slesru
+
+2012-02-13 17:24 +0000 [r354959]  Richard Mudgett <rmudgett at digium.com>
+
+	* res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
+	  reconnecting to pgsql database after connection loss. There can
+	  only be one database connection in res_config_pgsql just like
+	  res_config_sqlite. If the connection is lost, the connection may
+	  not get reestablished to the same database if the res_pgsql.conf
+	  and extconfig.conf files are inconsistent. * Made only use the
+	  configured database from res_pgsql.conf. * Fixed potential buffer
+	  overwrite of last[] in config_pgsql(). (closes issue
+	  ASTERISK-16982) Reported by: german aracil boned Review:
+	  https://reviewboard.asterisk.org/r/1731/ ........ Merged
+	  revisions 354953 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 19:51 +0000 [r355010]  Joshua Colp <jcolp at digium.com>
+
+	* /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
+	  at a time as otherwise they would share the same common local
+	  context list. (closes issue AST-758) ........ Merged revisions
+	  355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-13 22:03 +0000 [r355057]  Richard Mudgett <rmudgett at digium.com>
+
+	* pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
+	  execution. Since the dir timestamp is available at one second
+	  resolution, we cannot know if it was updated within the same
+	  second after we scanned it. Therefore, we will force another scan
+	  if the dir was just modified. * Changed to force another scan if
+	  the directory was just modified. (closes issue ASTERISK-19081)
+	  Reported by: Knut Bakke Review:
+	  https://reviewboard.asterisk.org/r/1688/ ........ Merged
+	  revisions 355056 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 09:49 +0000 [r355137]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c, /: call manager_event only if there is not
+	  null channel structure (Closes issue ASTERISK-19298) Reported by:
+	  robinfood Patches: issue19298.patch uploaded by may213 (License
+	  #5415) ........ Merged revisions 355136 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 13:33 +0000 [r355183]  Sean Bright <sean at malleable.com>
+
+	* /, channels/chan_iax2.c: Clear the high order bit from the
+	  destination call number before sending. send_apathetic_reply
+	  takes the incoming frame's source call number as the destination
+	  call number for the outgoing frame. If the incoming frame was a
+	  full frame, then the high order bit of the source call number is
+	  set and will be interpreted as a retransmit when sent back out as
+	  the destination call number. ........ Merged revisions 355182
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 15:53 +0000 [r355229]  Jason Parker <jparker at digium.com>
+
+	* /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
+	  CDRs by default in sample configs. ........ Merged revisions
+	  355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 16:27 +0000 [r355271]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Properly invert the return of a strncmp
+	  call. This was causing identification that should have been made
+	  private to be public. (closes issue AST-814) reported by Patrick
+	  Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
+	  (license 5430) ........ Merged revisions 355268 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-14 18:14 +0000 [r355375-355320]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
+	  in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
+	  Reported by: Alex Villacis Lasso Patches:
+	  asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
+	  (license #5617) patch uploaded by Alex Villacis Lasso Review:
+	  https://reviewboard.asterisk.org/r/1740/ ........ Merged
+	  revisions 355319 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  formats/format_ogg_vorbis.c: Fix voicemail problems when using
+	  ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
+	  format because it did not implement the seek and tell format
+	  callbacks among other problems. Since we were already using the
+	  libvorbis and libvorbisenc libraries we can use libvorbisfile as
+	  it is also part of the vorbis library package. * Made use the
+	  libvorbisfile to handle the ogg/vorbis file stream. The
+	  format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
+	  (closes issue ASTERISK-16926) Reported by: sque Patches:
+	  ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
+	  by sque ........ Merged revisions 355365 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-15 17:25 +0000 [r355530-355449]  Sean Bright <sean at malleable.com>
+
+	* /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
+	  intended. Back in r646, TRUNK_CALL_START was added and defined as
+	  0x4000. That same value was also hard-coded in one part of the
+	  IAX2 code instead of using the #define. TRUNK_CALL_START has
+	  changed over the years (for dealing with LOW_MEMORY), but the
+	  hard-coded usage was never updated to match. This patch fixes
+	  that. ........ Merged revisions 355448 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_iax2.c: Only use maxtrunkcall and
+	  maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These
+	  variables are only accessed from the IAX_OLD_FIND path, so there
+	  is no reason to keep them updated otherwise. ........ Merged
+	  revisions 355458 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
+	  sure to log 'apathetic' messages too. ........ Merged revisions
+	  355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-02-16 18:32 +0000 [r355620-355575]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, res/res_monitor.c: Fix AMI Monitor action without File header
+	  converting channel name into filename. * Fix potential Solaris
+	  crash if Monitor application has a urlbase and no fname_base
+	  option. ........ Merged revisions 355574 from

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