[asterisk-commits] bebuild: tag certified-1.8.11-cert5-rc1 r369850 - /certified/tags/1.8.11-cert...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Jul 9 15:01:04 CDT 2012


Author: bebuild
Date: Mon Jul  9 15:01:00 2012
New Revision: 369850

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369850
Log:
Importing files for 1.8.11-cert5-rc1 release.

Added:
    certified/tags/1.8.11-cert5-rc1/.lastclean   (with props)
    certified/tags/1.8.11-cert5-rc1/.version   (with props)
    certified/tags/1.8.11-cert5-rc1/ChangeLog   (with props)

Added: certified/tags/1.8.11-cert5-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert5-rc1/.lastclean?view=auto&rev=369850
==============================================================================
--- certified/tags/1.8.11-cert5-rc1/.lastclean (added)
+++ certified/tags/1.8.11-cert5-rc1/.lastclean Mon Jul  9 15:01:00 2012
@@ -1,0 +1,3 @@
+39
+
+

Propchange: certified/tags/1.8.11-cert5-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: certified/tags/1.8.11-cert5-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: certified/tags/1.8.11-cert5-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: certified/tags/1.8.11-cert5-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert5-rc1/.version?view=auto&rev=369850
==============================================================================
--- certified/tags/1.8.11-cert5-rc1/.version (added)
+++ certified/tags/1.8.11-cert5-rc1/.version Mon Jul  9 15:01:00 2012
@@ -1,0 +1,1 @@
+1.8.11-cert5-rc1

Propchange: certified/tags/1.8.11-cert5-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: certified/tags/1.8.11-cert5-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: certified/tags/1.8.11-cert5-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: certified/tags/1.8.11-cert5-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/certified/tags/1.8.11-cert5-rc1/ChangeLog?view=auto&rev=369850
==============================================================================
--- certified/tags/1.8.11-cert5-rc1/ChangeLog (added)
+++ certified/tags/1.8.11-cert5-rc1/ChangeLog Mon Jul  9 15:01:00 2012
@@ -1,0 +1,38423 @@
+2012-07-09  Asterisk Development Team
+
+	* Certified Asterisk 1.8.11-cert5-rc1 Released.
+
+2012-07-09 19:59 +0000 [r369848]  Matthew Jordan <mjordan at digium.com>
+
+	* channels/chan_iax2.c, main/channel.c, channels/chan_dahdi.c,
+	  channels/sig_analog.c, /, channels/chan_sip.c,
+	  include/asterisk/channel.h: Fix deadlock between bridged channels
+	  that attempt to set the hangup source Calling
+	  ast_set_hangupsource with the channel lock held can result in a
+	  deadlock because the function also locks the bridged channel.
+	  (issue AST-891)
+
+2012-07-09 19:50 +0000 [r369845]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Add support for exposing the received
+	  contact URI and also for setting the request URI in messages.
+	  (closes issue AST-911)
+
+2012-07-09 19:06 +0000 [r369839-369840]  Jason Parker <jparker at digium.com>
+
+	* include/asterisk/app_voicemail.h (removed): Remove file that
+	  should no longer exist.
+
+	* apps/app_mixmonitor.c, apps/app_voicemail.c,
+	  include/asterisk/callerid.h, include/asterisk/app.h,
+	  channels/chan_sip.c, apps/app_voicemail.exports.in,
+	  tests/test_voicemail_api.c, main/callerid.c, main/app.c,
+	  include/asterisk/app_voicemail.h: Re-merge changes that were
+	  reverted.
+	  ------------------------------------------------------------------------
+	  r365395 | qwell | 2012-05-04 16:17:08 -0500 (Fri, 04 May 2012) |
+	  7 lines Add support for folders in MixMonitor 'm' option.
+	  Backport manager actions. The manager actions are needed, so
+	  MixMonitor can be executed on existing channels. (issue DPMA-68)
+	  ------------------------------------------------------------------------
+	  r364761 | qwell | 2012-05-01 12:25:14 -0500 (Tue, 01 May 2012) |
+	  6 lines Remove folder_dir from voicemail snapshots API. It was
+	  both unused (except in tests, where it was fudged) and
+	  unnecessary. (closes issue AST-842)
+	  ------------------------------------------------------------------------
+	  r367161 | mmichelson | 2012-05-21 14:05:52 -0500 (Mon, 21 May
+	  2012) | 21 lines Add "send to voicemail" Digium phone
+	  functionality to Asterisk. This change accommodates two methods
+	  by which calls can be directed to a user's voicemail. * Incoming
+	  calls can be redirected to any user's voicemail. * Established
+	  calls can be blind transferred to any user's voicemail. Digium
+	  phones indicate the desire to direct a call to voicemail by using
+	  a Diversion header with a reason parameter of "send_to_vm". This
+	  patch adds the "send_to_vm" reason as a valid redirecting reason.
+	  In addition, chan_sip.c has been modified to update redirecting
+	  information on the transferred channel by reading a Diversion
+	  header on a REFER request. (closes issue AST-871) Reported by
+	  Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
+	  ------------------------------------------------------------------------
+	  r368790 | mjordan | 2012-06-12 08:44:36 -0500 (Tue, 12 Jun 2012)
+	  | 18 lines Fix deadlock in SIP transfers that involve a REFER
+	  request In r367163, "send to voicemail" functionality was added
+	  to the SIP channel driver. This required updating the party
+	  redirecting information for the channel based on the headers
+	  provided in the REFER request. When the redirecting party
+	  information is updated on the channel, a call to
+	  ast_indicate_data occurs. Because handle_request_refer still had
+	  the sip_pvt locked, a deadlock could occur between the pbx_thread
+	  and the do_monitor thread servicing the REFER request. This patch
+	  preserves the proper locking order between the channel and the
+	  sip_pvt by ensuring that the sip_pvt is unlocked prior to
+	  updating the party redirecting information on the channel.
+	  (closes issue AST-903) Reported by: Matt Jordan patches:
+	  jira_ast_903_trunk.patch by rmudgett (license 5621)
+	  ------------------------------------------------------------------------
+	  r368962 | qwell | 2012-06-14 13:38:48 -0500 (Thu, 14 Jun 2012) |
+	  11 lines Remove global symbol requirement from app_voicemail.
+	  This uses the existing "function installation" stuff that already
+	  existed for other functions, like getting message counts. (closes
+	  issue AST-807) (issue AST-901) (issue AST-908) Review:
+	  https://reviewboard.asterisk.org/r/1965/
+	  ------------------------------------------------------------------------
+	  r368964 | qwell | 2012-06-14 14:03:24 -0500 (Thu, 14 Jun 2012) |
+	  8 lines These functions that were moved need to be static. Also
+	  wrap test functions in a #ifdef. (issue AST-807) (issue AST-901)
+	  (issue AST-908)
+	  ------------------------------------------------------------------------
+	  r368998 | qwell | 2012-06-15 10:31:43 -0500 (Fri, 15 Jun 2012) |
+	  6 lines Remove some symbol exports that got missed in the removal
+	  of global symbols. (issue AST-807) (issue AST-901) (issue
+	  AST-908)
+	  ------------------------------------------------------------------------
+	  r369024 | qwell | 2012-06-15 11:29:40 -0500 (Fri, 15 Jun 2012) |
+	  2 lines Fix voicemail API tests by using the correct argument
+	  order for create/destroy.
+	  ------------------------------------------------------------------------
+
+2012-07-05  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert4 Released.
+
+	* AST-2012-010
+
+	* AST-2012-011
+
+2012-05-29  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert2 Released.
+
+	* AST-2012-007
+
+	* AST-2012-008
+
+2012-04-25  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Certified Asterisk 1.8.11-cert1 Released.
+
+2012-04-25 16:53 +0000 [r363674]  Jason Parker <jparker at digium.com>
+
+	* / (added): Asterisk 1.8-digiumphones branch has become Certified
+	  Asterisk 1.8.11. For more details about Certified Asterisk, see
+	  http://tinyurl.com/7pfp639
+
+2012-04-24 20:57 +0000 [r363374]  Jason Parker <jparker at digium.com>
+
+	* /res/res_smdi.c,
+	  /apps/app_osplookup.c,
+	  /channels/chan_misdn.c,
+	  /channels/chan_skinny.c,
+	  /funcs/func_frame_trace.c,
+	  /cdr/cdr_sqlite.c,
+	  /pbx/pbx_realtime.c,
+	  /apps/app_amd.c,
+	  /pbx/pbx_dundi.c,
+	  /apps/app_url.c,
+	  /channels/chan_nbs.c,
+	  /apps/app_externalivr.c,
+	  /apps/app_zapateller.c,
+	  /cdr/cdr_odbc.c,
+	  /res/res_fax_spandsp.c,
+	  /channels/chan_mgcp.c,
+	  /cel/cel_pgsql.c,
+	  /apps/app_readfile.c,
+	  /apps/app_test.c,
+	  /apps/app_ices.c,
+	  /channels/chan_gtalk.c,
+	  /cdr/cdr_csv.c,
+	  /channels/chan_phone.c,
+	  /funcs/func_pitchshift.c,
+	  /apps/app_waitforring.c,
+	  /formats/format_vox.c,
+	  /res/res_timing_pthread.c,
+	  /apps/app_minivm.c,
+	  /channels/chan_h323.c,
+	  /cel/cel_sqlite3_custom.c,
+	  /apps/app_confbridge.c,
+	  /res/res_config_ldap.c,
+	  /apps/app_nbscat.c,
+	  /cdr/cdr_sqlite3_custom.c,
+	  /res/res_snmp.c,
+	  /apps/app_dictate.c,
+	  /apps/app_waitforsilence.c,
+	  /apps/app_dahdiras.c,
+	  /pbx/pbx_lua.c,
+	  /apps/app_alarmreceiver.c,
+	  /apps/app_image.c,
+	  /res/res_ael_share.c,
+	  /cdr/cdr_tds.c,
+	  /apps/app_setcallerid.c,
+	  /apps/app_mp3.c,
+	  /channels/chan_alsa.c,
+	  /res/res_timing_kqueue.c,
+	  /channels/chan_unistim.c,
+	  /apps/app_dahdibarge.c,
+	  /res/res_config_pgsql.c,
+	  /res/res_adsi.c,
+	  /res/res_phoneprov.c,
+	  /apps/app_morsecode.c,
+	  /cdr/cdr_pgsql.c,
+	  /res/res_config_sqlite.c,
+	  /channels/chan_jingle.c,
+	  /pbx/pbx_ael.c,
+	  /apps/app_sms.c,
+	  /formats/format_jpeg.c,
+	  /apps/app_jack.c,
+	  /apps/app_adsiprog.c,
+	  /cel/cel_radius.c,
+	  /res/res_ais.c,
+	  /cel/cel_tds.c,
+	  /apps/app_festival.c,
+	  /apps/app_chanisavail.c,
+	  /channels/chan_console.c,
+	  /apps/app_talkdetect.c,
+	  /res/res_jabber.c,
+	  /cdr/cdr_radius.c,
+	  /apps/app_getcpeid.c,
+	  /channels/chan_oss.c: Disable extended
+	  and deprecated modules by default. Users can still enable any of
+	  these using menuselect if they so choose. (closes issue AST-873)
+
+2012-04-23 15:17 +0000 [r363161]  Jason Parker <jparker at digium.com>
+
+	* /main/manager.c,
+	  ,
+	  /channels/chan_sip.c,
+	  /channels/chan_skinny.c: Multiple
+	  revisions 363102,363106,363141 ........ r363102 | mjordan |
+	  2012-04-23 08:37:55 -0500 (Mon, 23 Apr 2012) | 16 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 ........
+	  r363106 | mjordan | 2012-04-23 09:05:02 -0500 (Mon, 23 Apr 2012)
+	  | 17 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) ........ r363141 | jrose | 2012-04-23 09:33:16 -0500 (Mon,
+	  23 Apr 2012) | 20 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 363102,363106,363141 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-04-19 20:31 +0000 [r362673]  Mark Michelson <mmichelson at digium.com>
+
+	* /channels/chan_sip.c: Add a test
+	  application for sending custom SIP INFO messages. When
+	  TEST_FRAMEWORK is enabled, SIPSendCustomInfo is available to test
+	  sending custom INFO requests. Review:
+	  https://reviewboard.asterisk.org/r/1866
+
+2012-04-13 17:19 +0000 [r362042-362132]  Matthew Jordan <mjordan at digium.com>
+
+	* : Rename property branches-1.8-merged to
+	  branch-1.8-merged
+
+	* : Update properties on 1.8-digiumphones
+	  Change the merge property tag from svnmerge-integrated to
+	  branches-1.8-merged. Added merged revisions from r362042.
+
+	* ,
+	  /channels/chan_sip.c,
+	  /main/features.c: Merge of several
+	  needed fixes for 1.8-digiumphones This merges fixes for the
+	  following issues into the 1.8-digiumphones branch: *
+	  ASTERISK-19355 - Call transfer with consultation frequently fails
+	  in cross- linked Asterisk scenario (directmedia & sendrpid
+	  active) * ASTERISK 19365 - Remote SIP Call legs are frequently
+	  not released in a cross-linked Asterisk scenario (directmedia &
+	  sendrpid) * ASTERISK-19183 - Sporadically missing connectedline
+	  event to caller channel in directed pickup app
+
+2012-04-09 20:40 +0000 [r361704]  Mark Michelson <mmichelson at digium.com>
+
+	* /apps/app_voicemail.c,
+	  /apps/app_voicemail.exports.in,
+	  /tests/test_voicemail_api.c (added),
+	  /include/asterisk/app_voicemail.h: Fix
+	  bugs in voicemail APIs and add unit tests. There were several
+	  crashes that could occur due to NULL inputs, invalid inputs, and
+	  the like. This fixes all known ones and adds unit tests to
+	  exercise the APIs.
+
+2012-04-06 19:08 +0000 [r361502]  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 17:24 +0000 [r361283]  Mark Michelson <mmichelson at digium.com>
+
+	* /funcs/func_presence_state.c,
+	  /tests/test_config.c: Add additional
+	  configuration and presence unit tests. These were originally
+	  written while merging features into trunk, but these tests apply
+	  just as much for the 1.8 version of Digium phones, so might as
+	  well have them here, too.
+
+2012-04-03 21:03 +0000 [r361088]  Jonathan Rose <jrose at digium.com>
+
+	* /apps/app_mixmonitor.c: Make m option
+	  for mixmonitor delete the source file once it is finished copying
+	  to vm. Review: https://reviewboard.asterisk.org/r/1842/
+
+2012-03-29 21:49 +0000 [r360826]  Jason Parker <jparker at digium.com>
+
+	* /main/manager.c,
+	  ,
+	  /main/utils.c,
+	  /include/asterisk/manager.h,
+	  /apps/app_milliwatt.c: Multiple
+	  revisions 359656,359706,359979 ........ r359656 | mjordan |
+	  2012-03-15 13:35:59 -0500 (Thu, 15 Mar 2012) | 22 lines Fix
+	  remotely exploitable stack overrun in Milliwatt Milliwatt is
+	  vulnerable to a remotely exploitable stack overrun when using the
+	  'o' option. This occurs due to the milliwatt_generate function
+	  not accounting for AST_FRIENDLY_OFFSET when calculating the
+	  maximum number of samples it can put in the output buffer. This
+	  patch resolves this issue by taking into account
+	  AST_FRIENDLY_OFFSET when determining the maximum number of
+	  samples allowed. Note that at no point is remote code execution
+	  possible. The data that is written into the buffer is the
+	  pre-defined Milliwatt data, and not custom data. (closes issue
+	  ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt
+	  Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell
+	  Bryant (license 6283) Note that this patch was written by
+	  Russell, even though Matt uploaded it ........ Merged revisions
+	  359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
+	  ........ r359706 | mjordan | 2012-03-15 14:01:22 -0500 (Thu, 15
+	  Mar 2012) | 16 lines Fix remotely exploitable stack overflow in
+	  HTTP manager There exists a remotely exploitable stack buffer
+	  overflow in HTTP digest authentication handling in Asterisk. The
+	  particular method in question is only utilized by HTTP AMI. When
+	  parsing the digest information, the length of the string is not
+	  checked when it is copied into temporary buffers allocated on the
+	  stack. This patch fixes this behavior by parsing out pre-defined
+	  key/value pairs and avoiding unnecessary copies to the stack.
+	  (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
+	  by: Matt Jordan ........ r359979 | rmudgett | 2012-03-20 12:21:16
+	  -0500 (Tue, 20 Mar 2012) | 28 lines Allow AMI action callback to
+	  be reentrant. Fix AMI module reload deadlock regression from
+	  ASTERISK-18479 when it tried to fix the race between calling an
+	  AMI action callback and unregistering that action. Refixes
+	  ASTERISK-13784 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 patch 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/ Review:
+	  https://reviewboard.asterisk.org/r/1820/ ........ Merged
+	  revisions 359656,359706,359979 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-21 15:44 +0000 [r360031-360188]  Mark Michelson <mmichelson at digium.com>
+
+	* /main/pbx.c: Prevent potentially passing
+	  a NULL pointer to strcasecmp()
+
+	* /main/pbx.c: Fix one more "(null)"
+	  string. If a hint with no presence portion were added, it would
+	  result in another "(null)" string warning.
+
+	* /main/pbx.c: Fix another "Possible
+	  programming error" bug. Similar to the previous commit, don't
+	  pass a printf-generated string to ast_strlen_zero.
+
+	* /main/pbx.c: Get rid of an annoying
+	  "Possible programming error" message. If an extension's 'app'
+	  field is NULL, then a "(null)" string would be written into an
+	  ast_str due to the way that snprintf works. When this is passed
+	  to ast_strlen_zero(), it fires up a big warning indicating
+	  something is probably wrong. There indeed was a problem, but
+	  luckily it wasn't a very big problem. After the failed
+	  ast_strlen_zero() check and big warning message, the very next if
+	  statement, checking to see if the "(null)" matched a presence
+	  provider, would fail, so no harm was done.
+
+2012-03-08 18:40 +0000 [r358725]  Jonathan Rose <jrose at digium.com>
+
+	* /apps/app_mixmonitor.c: Fixes
+	  unitialized variable use warning introduced by addition of
+	  mixmonitor forward to vm
+
+2012-03-08 18:02 +0000 [r358692]  Jason Parker <jparker at digium.com>
+
+	* , /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. (closes issue
+	  ASTERISK-19430) ........ Merged revisions 357665 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2012-03-02 15:18 +0000 [r357808]  Paul Belanger <pabelanger at digium.com>
+
+	* /apps/app_mixmonitor.c: Fixed xmldoc
+	  formatting error for 'm' option
+
+2012-02-28 21:52 +0000 [r357456-357459]  Jason Parker <jparker at digium.com>
+
+	* /main/channel.c,
+	  /funcs/func_presence_state.c (added),
+	  /main/manager.c,
+	  /channels/chan_skinny.c,
+	  /funcs/func_frame_trace.c,
+	  /include/asterisk/jabber.h,
+	  /main/file.c,
+	  /main/app.c,
+	  /tests/test_config.c (added),
+	  /include/asterisk/frame.h,
+	  /main/custom_control_frame.c (added),
+	  /main/message.c (added),
+	  /apps/app_mixmonitor.c,
+	  /channels/sip/include/sip.h,
+	  /main/asterisk.c,
+	  /tests/test_custom_control.c (added),
+	  /main/pbx.c,
+	  /include/asterisk/presencestate.h
+	  (added),
+	  /include/asterisk/app_voicemail.h
+	  (added), /include/asterisk/channel.h,
+	  /include/asterisk/manager.h,
+	  /apps/app_queue.c,
+	  /main/config.c,
+	  /include/asterisk/file.h,
+	  /include/asterisk/app.h,
+	  /include/asterisk/event_defs.h,
+	  /configs/jabber.conf.sample,
+	  /include/asterisk/custom_control_frame.h
+	  (added), /include/asterisk/message.h
+	  (added), /main/features.c,
+	  /apps/app_voicemail.exports.in,
+	  /main/event.c,
+	  /include/asterisk/pbx.h,
+	  /configs/sip.conf.sample,
+	  /apps/app_voicemail.c,
+	  /channels/chan_sip.c,
+	  /include/asterisk/config.h,
+	  /configs/manager.conf.sample,
+	  /include/asterisk/_private.h,
+	  /res/res_jabber.c,
+	  /main/presencestate.c (added): Add
+	  support for Digium Phones.
+
+
+2012-03-29  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.11.0 Released.
+
+2012-03-26  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.8.11.0-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 tried
+	  to fix the race between calling an AMI action callback and
+	  unregistering that action.  Refixes ASTERISK-13784 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 patch 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 1.8.11.0-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 21:57 +0000 [r353368-353320]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/sip/include/sip.h, channels/sip/include/dialog.h,
+	  channels/chan_sip.c: 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: prevent debug messsges displaying -ve Cseq
+	  numbers. Missed in R353320
+
+2012-01-30 23:17 +0000 [r353371]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/dnsmgr.h, main/dnsmgr.c, channels/chan_sip.c:
+	  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/
+
+2012-01-31 16:51 +0000 [r353454]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/channel.h, main/manager.c: 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/
+
+2012-01-31 23:41 +0000 [r353502]  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/
+
+2012-02-01 15:02 +0000 [r353550]  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
+
+2012-02-01 15:50 +0000 [r353598]  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
+
+2012-02-01 21:05 +0000 [r353769-353720]  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)
+
+	* 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)
+
+2012-02-02 16:58 +0000 [r353770]  Mark Michelson <mmichelson at digium.com>
+
+	* UPGRADE.txt, configs/manager.conf.sample,
+	  include/asterisk/manager.h, configs/http.conf.sample,
+	  main/manager.c, main/http.c: 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
+
+2012-02-02 18:31 +0000 [r353818]  Jonathan Rose <jrose at digium.com>
+
+	* funcs/func_curl.c: Backports some documentation for func_curl
+	  from 10 to 1.8 For some reason this function was completely
+	  undocumented in 1.8. I copied the 10 docs over to 1.8 and removed
+	  references to an enumerator that was added in the Asterisk 10
+	  version of func_curl. That was the only change I noted. (closes
+	  issue ASTERISK-19186) Reported by: Olivier Krief
+
+2012-02-02 20:01 +0000 [r353867]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/sig_pri.c, channels/sig_pri.h, channels/chan_dahdi.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
+
+2012-02-02 22:26 +0000 [r353915]  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
+
+2012-02-03 21:24 +0000 [r353999]  Jonathan Rose <jrose at digium.com>
+
+	* 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/
+
+2012-02-06 17:28 +0000 [r354216-354116]  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
+
+	* 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
+
+2012-02-07 15:04 +0000 [r354263]  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/
+
+2012-02-07 20:53 +0000 [r354348]  Terry Wilson <twilson at digium.com>
+
+	* contrib/realtime/postgresql/realtime.sql, channels/chan_sip.c:
+	  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/
+
+2012-02-09 02:23 +0000 [r354492]  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/
+
+2012-02-09 02:52 +0000 [r354495]  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
+
+2012-02-09 16:30 +0000 [r354542]  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
+
+2012-02-09 16:56 +0000 [r354545]  Mark Michelson <mmichelson at digium.com>
+
+	* CHANGES, 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
+
+2012-02-09 17:07 +0000 [r354547]  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/
+
+2012-02-09 17:32 +0000 [r354640-354594]  Mark Michelson <mmichelson at digium.com>
+
+	* main/translate.c: Fix translation path choices. This change makes
+	  it so computational cost is not taken into account when deciding
+	  if a multistep path is better than a single-step path. This means
+	  that the only time a multistep path will be chosen is if no
+	  single-step path exists. This ensures a better quality
+	  translation even if it turns out to be slightly slower. (closes
+	  issue ASTERISK-16821) reported by Andrew Lindh Review:
+	  https://reviewboard.asterisk.org/r/1715
+
+	* main/translate.c: Remove outdated comment.
+

[... 37642 lines stripped ...]



More information about the asterisk-commits mailing list