[asterisk-commits] bebuild: tag 10.0.0-rc1 r343795 - /tags/10.0.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Nov 8 07:35:59 CST 2011


Author: bebuild
Date: Tue Nov  8 07:35:55 2011
New Revision: 343795

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=343795
Log:
Importing files for 10.0.0-rc1 release.

Added:
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    tags/10.0.0-rc1/ChangeLog   (with props)

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+2011-11-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 10.0.0-rc1 Released.
+
+2011-11-08 13:26 +0000 [r343789-343792]  Leif Madsen <lmadsen at digium.com>
+
+	* /: Recorded merge of revisions 343791 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Fix
+	  boo-boo in prep_tarball script. A hardcoded a branch number was
+	  in the prep_tarball which could not work. Changed it to the
+	  variable.
+
+	* build_tools/prep_tarball: Fix boo-boo in prep_tarball script. A
+	  hardcoded a branch number was in the prep_tarball which could not
+	  work. Changed it to the variable.
+
+2011-11-07 22:37 +0000 [r343743]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Make "sip show settings" CLI command get
+	  RPID flags from the right global page The "Trust RPID" and "Send
+	  RPID" entries in the "sip show settings" CLI command pulled the
+	  flags from the incorrect global flags page. These are now read
+	  from sip global flags page 0. (closes issue AST-711)
+
+2011-11-07 21:42 +0000 [r343691]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: respect case changes in peer names on sip
+	  reload ASTERISK-18669 ........ Merged revisions 343690 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 21:27 +0000 [r343677]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
+	  changing dialogs hash key callid. Changing an object value used
+	  as a container key requires removing the object from the
+	  container and reinserting it. * Created change_callid_pvt() to
+	  call instead of build_callid_pvt(). The change_callid_pvt() will
+	  correctly change the dialog callid so the ao2 conainter can
+	  explicitly unlink it. ........ Merged revisions 343637 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 20:31 +0000 [r343635]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: Prevent BLF subscriptions from causing
+	  deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
+	  was causing deadlocks. This function now requires that both the
+	  peer and associated pvt be unlocked before it is called for cases
+	  where peer and peer->mwipvt form a circular reference. (closes
+	  issue ASTERISK-18663) Review:
+	  https://reviewboard.asterisk.org/r/1563/ ........ Merged
+	  revisions 343621 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 19:55 +0000 [r343580]  wdoekes <wdoekes at localhost>:
+
+	* main/udptl.c, UPGRADE.txt: Correct the default udptl port range.
+	  The udptl port range was defined as 4000-4999 in the
+	  udptl.conf.sample, as 4500-4599 if you didn't have a config and
+	  4500-4999 if your config was broken. Default is now 4000-4999.
+	  (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher
+	  Review: https://reviewboard.asterisk.org/r/1565
+
+2011-11-07 19:51 +0000 [r343578]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_sip.c: Fix deadlock if peer is destroyed while
+	  sending MWI notice. A dialog cannot be destroyed by the
+	  ao2_callback dialog_needdestroy because of a deadlock between the
+	  dialogs container lock and the RWLOCK of the events subscription
+	  list. * Create dialogs_to_destroy container to hold dialogs that
+	  will be destroyed. * Ensure that the event subscription callback
+	  will never happen with an invalid peer pointer by making the
+	  event callback removal the first thing in the peer destructor
+	  callback. NOTE: This particular deadlock will not happen with
+	  Asterisk 10, but some of the changes still apply. (closes issue
+	  ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review:
+	  https://reviewboard.asterisk.org/r/1564/ ........ Merged
+	  revisions 343577 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-07 18:39 +0000 [r343533]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/format.c: list all of the codecs associated with a
+	  particular format id for CLI command "core show codec" AST-699
+
+2011-11-04 15:11 +0000 [r343445]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c,
+	  addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/dlist.c, /,
+	  addons/ooh323c/src/dlist.h, addons/ooh323c/src/printHandler.c:
+	  Final fix memleaks in GkClient codes, same for Timer codes.
+	  (these memleaks stop development of gk codes, now i can continue)
+	  Fix printHandler 'Unbalanced Structure' issues with locking
+	  printHandler data for single thread. ........ Merged revisions
+	  343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 20:31 +0000 [r343393]  wdoekes <wdoekes at localhost>:
+
+	* /, res/res_config_sqlite.c: Fix sqlite config driver segfault and
+	  broken queries The sqlite realtime handler assumed you had a
+	  static config configured as well. The realtime multientry handler
+	  assumed that you weren't using dynamic realtime. (closes issue
+	  ASTERISK-18354) (closes issue ASTERISK-18355) Review:
+	  https://reviewboard.asterisk.org/r/1561 ........ Merged revisions
+	  343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 19:57 +0000 [r343337]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, funcs/func_dialgroup.c: Remove invalid flag given to iterator
+	  in func_dialgroup.c ........ Merged revisions 343336 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-03 15:39 +0000 [r343221-343277]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/sip/include/sip.h: Make room for the fax detect flags
+	  The original REGISTERTRYING flag, in addition to being impossible
+	  to check, also encroached on the space for the flag above it.
+	  This patch moves the flags that were below REGISTERTRYING back to
+	  where they were as though we had just removed the REGISTERTRYING
+	  option. ........ Merged revisions 343276 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
+	  channels/sip/include/sip.h: Remove registertrying option in
+	  chan_sip This option is not only useless, but has been broken
+	  since inception since the flag was never copied from the peer
+	  where it is set to the pvt where it was checked. RFC 3261
+	  specificially states that you should not send a provisional
+	  response to a non-INVITE request, and if we did fix the code so
+	  that it worked, it would cause the same kind of user enumeration
+	  vulnerability that we've discussed with the nat= setting. This
+	  patch removes registertrying option and any code that would have
+	  sent a 100 response to a register. Review:
+	  https://reviewboard.asterisk.org/r/1562/ ........ Merged
+	  revisions 343220 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 22:24 +0000 [r343158-343192]  wdoekes <wdoekes at localhost>:
+
+	* /, channels/chan_sip.c: Fix improper warning introduced by
+	  r342927 and more tweaks Changeset r342927 introduced a warning
+	  which was only supposed to be emitted when a found realtime peer
+	  had an empty (or no) name. It turned out that there were some
+	  inconsistencies left. Now found peers with an empty name are
+	  explicitly ignored like before r342927 but better. Reviewed by:
+	  Stefan Schmidts, Terry Wilson Review:
+	  https://reviewboard.asterisk.org/r/1560 ........ Merged revisions
+	  343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* include/asterisk/stringfields.h, include/asterisk/utils.h, /,
+	  main/utils.c: Ensure that string field lengths are properly
+	  aligned Integers should always be aligned. For some platforms
+	  (ARM, SPARC) this is more important than for others. This
+	  changeset ensures that the string field string lengths are
+	  aligned on *all* platforms, not just on the SPARC for which there
+	  was a workaround. It also fixes that the length integer can be
+	  resized to 32 bits without problems if needed. (closes issue
+	  ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
+	  Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
+	  https://reviewboard.asterisk.org/r/1549 ........ Merged revisions
+	  343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 19:33 +0000 [r343048-343103]  Leif Madsen <lmadsen at digium.com>
+
+	* /, apps/app_authenticate.c: Add note about how Authenticate()
+	  application with option 'd' works. (closes issue ASTERISK-17422)
+	  Reported by: Leif Madsen ........ Merged revisions 343102 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, configs/queues.conf.sample: Update documentation for
+	  leastrecent strategy. In queues.conf.sample the leastrecent
+	  strategy was incorrectly described. Now updated to reflect how
+	  the strategy actually checks peers. (closes issue ASTERISK-17854)
+	  Reported by: Sebastian Denz Patches: queues.conf-doc_issue.patch
+	  (License #6139) ........ Merged revisions 343047 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-02 13:45 +0000 [r342991]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, apps/app_meetme.c: Modify comments in MeetMe application
+	  documentation about DAHDI. The MeetMe application documentation
+	  has some comments about usage of DAHDI, and they were a bit
+	  outdated relative to modern DAHDI releases. This patch changes
+	  the comment to just tell the user that a functional DAHDI timing
+	  source is required, and no longer mention 'dahdi_dummy', since
+	  that module does not exist in current DAHDI releases. ........
+	  Merged revisions 342990 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-11-01 20:58 +0000 [r342870-342929]  wdoekes <wdoekes at localhost>:
+
+	* /, channels/chan_sip.c, configs/extconfig.conf.sample,
+	  include/asterisk/config.h, main/config.c: Several fixes to the
+	  chan_sip dynamic realtime peer/user lookup There were several
+	  problems with the dynamic realtime peer/user lookup code. The
+	  lookup logic had become rather hard to read due to lots of
+	  incremental changes to the realtime_peer function. And, during
+	  the addition of the sipregs functionality, several possibilities
+	  for memory leaks had been introduced. The insecure=port matching
+	  has always been broken for anyone using the sipregs family. And,
+	  related, the broken implementation forced those using sipregs to
+	  *still* have an ipaddr column on their sippeers table. Thanks
+	  Terry Wilson for comprehensive testing and finding and fixing
+	  unexpected behaviour from the multientry realtime call which
+	  caused the realtime_peer to have a completely unused code path.
+	  This changeset fixes the leaks, the lookup inconsistenties and
+	  that you won't need an ipaddr column on your sippeers table
+	  anymore (when you're using sipregs). Beware that when you're
+	  using sipregs, peers with insecure=port will now start matching!
+	  (closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
+	  Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
+	  Wilson Review: https://reviewboard.asterisk.org/r/1395 ........
+	  Merged revisions 342927 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* contrib/realtime/mysql/sipfriends.sql (removed),
+	  contrib/realtime/mysql/sippeers.sql (added),
+	  configs/res_config_mysql.conf.sample, /,
+	  configs/extconfig.conf.sample, configs/res_ldap.conf.sample,
+	  res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample,
+	  main/config.c: Cleanup references to sipusers and sipfriends
+	  dynamic realtime families Somewhere between 1.4 and 1.8 the
+	  sipusers family has become completely unused. Before that, the
+	  sipfriends family had been obsoleted in favor of separate
+	  sipusers and sippeers families. Apparently, they have been merged
+	  back again into a single family which is now called "sippeers".
+	  Reviewed by: irroot, oej, pabelanger Review:
+	  https://reviewboard.asterisk.org/r/1523 ........ Merged revisions
+	  342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-31 17:46 +0000 [r342824]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/format.c, main/format_cap.c: Misc format capability fixes. *
+	  Fixed typo in format_cap.c:joint_copy_helper() using the wrong
+	  variable. * Fix potential race between checking if an interface
+	  exists and adding it to the container in
+	  format.c:ast_format_attr_reg_interface(). * Fixed double rwlock
+	  destroy in format.c:ast_format_attr_init() error exit path. *
+	  Simplified format.c:find_interface() and
+	  format.c:has_interface().
+
+2011-10-31 16:04 +0000 [r342770]  Matthew Jordan <mjordan at digium.com>
+
+	* main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access
+	  when adding extension to pattern match tree When an extension is
+	  removed from a context, its entry in the pattern match tree is
+	  not deleted. Instead, the extension is marked as deleted. When an
+	  extension is removed and re-added, if that extension is also a
+	  prefix of another extension, several log messages would report an
+	  error and did not check whether or not the extension was deleted
+	  before accessing the memory. Additionally, if the extension was
+	  already in the tree but previously deleted, and the pattern was
+	  at the end of a match, the findonly flag was not honored and the
+	  extension would be erroneously undeleted. Additionaly, it was
+	  discovered that an IAX2 peer could be unregistered via the CLI,
+	  while at the same time it could be scheduled for unregistration
+	  by Asterisk. The unregistration method now checks to see if the
+	  peer was already unregistered before continuing with an
+	  unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
+	  Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
+	  Review: https://reviewboard.asterisk.org/r/1526 ........ Merged
+	  revisions 342769 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-30 02:21 +0000 [r342715]  Terry Wilson <twilson at digium.com>
+
+	* res/res_calendar.c: Don't crash on empty notify channel
+
+2011-10-29 04:26 +0000 [r342662]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix
+	  AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
+	  AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
+	  iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
+	  the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
+	  list if AST_LIST_INSERT_BEFORE_CURRENT() or
+	  AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
+	  cut and paste error using the wrong variable in
+	  AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
+	  for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
+	  AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-27 20:10 +0000 [r342605]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/dsp.c: tweak the v21 detector to detect an additional
+	  pattern of hits and misses
+
+2011-10-27 19:41 +0000 [r342546-342603]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_rtp_multicast.c, /: Fix sequence number overflow over 16
+	  bits causing codec change in RTP packets. Sequence number was
+	  handled as an unsigned integer (usually 32 bits I think, more
+	  depending on the architecture) and was put into the rtp packet
+	  which is basically just a bunch of bits using an or operation.
+	  Sequence number only has 16 bits allocated to it in an RTP packet
+	  anyway, so it would add to the next field which just happened to
+	  be the codec. This makes sure the sequence number is set to be a
+	  16 bit integer regardless of architecture (hopefully) and also
+	  makes it so the incrementing of the sequence number does bitwise
+	  or at the peak of a 16 bit number so that the value will be set
+	  back to 0 when going beyond 65535 anyway. (closes issue
+	  ASTERISK-18291) Reported by: Will Schick Review:
+	  https://reviewboard.asterisk.org/r/1542/ ........ Merged
+	  revisions 342602 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, res/res_jabber.c: Cleanup reference leaks in res_jabber
+	  res_jabber.c had a number of places where astobjs would be
+	  referenced and have their reference counts bumped without having
+	  a dereference made before the object lost scope. This patch adds
+	  a number of ASTOBJ_UNREFs to resolve that. Review:
+	  https://reviewboard.asterisk.org/r/1478/ ........ Merged
+	  revisions 342545 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 22:05 +0000 [r342485-342488]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/astobj2.c: Check fopen return value for ao2 reference
+	  debug output. Reported by: wdoekes Patched by: wdoekes Review:
+	  https://reviewboard.asterisk.org/r/1539/ ........ Merged
+	  revisions 342487 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/sig_pri.c: Change D-channel warning to be less
+	  confusing on non-NFAS setups. The "No D-channels available! Using
+	  Primary channel as D-channel anyway!" WARNING message has been
+	  confusing on non-NFAS setups. The message refers to things that
+	  are NFAS specific. * Changed the warning to several different
+	  warnings to be more accurate for the situation and less confusing
+	  as a result: "No D-channels up! Switching selected D-channel from
+	  X to Y.", "No D-channels up!", and "D-channel is down!". ........
+	  Merged revisions 342484 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 21:10 +0000 [r342381-342436]  Terry Wilson <twilson at digium.com>
+
+	* /, apps/app_queue.c: Use int for storing ao2_container_count
+	  instad of size_t AST-676 ........ Merged revisions 342435 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_queue.c: Simplify queue membercount code Despite an
+	  ominous sounding comment stating that membercount was for "logged
+	  in" members only and thus we couldn't use ao2_container_count(),
+	  I could not find a single place in the code where that seemed to
+	  be accurate. The only time we decremented membercount was when we
+	  were marking something dead or actually removing it. The only
+	  places we incremented it were either after ao2_link(), or trying
+	  to correct for having set it to 0 during a reload. In every case
+	  where we were correcting the value, it seemed that we were trying
+	  to make the count actually match what ao2_container_count() would
+	  return. The only place I could find where we made a determination
+	  about something being "logged in" or not, we didn't trust the
+	  membercount, but instead looked at devicestate, paused, etc. This
+	  patch removes membercount, replaces its use with
+	  ao2_container_count, and manually adds the results of
+	  ao2_container_count to a "membercount" field for ast_data queue
+	  query results. This patch also would fix AST-676, but as it is
+	  slightly riskier than the previously committed fix, the two
+	  commits have been made separately. Reivew:
+	  https://reviewboard.asterisk.org/r/1541/ ........ Merged
+	  revisions 342383 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, apps/app_queue.c: Properly update membercount for reloaded
+	  members Since q->membercount is set to 0 before reloading, it is
+	  important to increment it again for reloaded members as well as
+	  added. (closes issue AST-676) Review:
+	  https://reviewboard.asterisk.org/r/1541/ ........ Merged
+	  revisions 342380 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-25 19:08 +0000 [r342277-342329]  Kinsey Moore <kmoore at digium.com>
+
+	* pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for
+	  pbx_spool.c One of the changes in the recent spool handling of
+	  hardlinks patch was just outside a HAVE_INOTIFY block and caused
+	  compilation to fail in some build environments. This has been
+	  corrected. ........ Merged revisions 342328 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* pbx/pbx_spool.c, /: Merged revisions 342276 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) |
+	  18 lines Fix spool handling to allow call files to be hardlinked
+	  into place This fixes the inotify code to handle call files being
+	  hardlinked into the spool directory. The smsq utility does this,
+	  instead of rename(), to ensure that it cannot accidentally
+	  overwrite an existing spool file. A rename() might do that, but
+	  link() will definitely not. The inotify code had broken this,
+	  because it would wait for an IN_CLOSE_WRITE event on the file...
+	  which was never forthcoming, since it was never opened. Now we
+	  look for IN_OPEN events following the IN_CREATE event, and only
+	  wait for an IN_CLOSE_WRITE if the file was actually opened.
+	  Patch-by: dwmw2 (closes issue ASTERISK-18331) Review:
+	  https://reviewboard.asterisk.org/r/1391/ ........
+
+2011-10-25 01:25 +0000 [r342224]  Terry Wilson <twilson at digium.com>
+
+	* /, include/asterisk/config.h, main/config.c: Return NULL when no
+	  results returned for realtime_multientry It was not documented
+	  what the return value should be when no entries were returned
+	  with the multientry realtime callback. This change forces
+	  consistent behavior even if the backends return an empty
+	  ast_config. Review: https://reviewboard.asterisk.org/r/1521/
+	  ........ Merged revisions 342223 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-24 22:32 +0000 [r342183]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add
+	  missing link/unlink nolock debug defines.
+
+2011-10-24 19:51 +0000 [r342062]  Jonathan Rose <jrose at digium.com>
+
+	* /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now
+	  include fromuser of related peer. This behavior matches up more
+	  closely with the way invite/register/etc are handled. This patch
+	  also modifies some adjacent code for code style compliance.
+	  Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
+	  Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
+	  by Jeremy Kister (license #6232) ........ Merged revisions 342061
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-24 07:31 +0000 [r341920-342017]  Gregory Nietsky <gregory at distrotech.co.za>
+
+	* apps/app_queue.c: queues container needs locking when using the
+	  OBJ_NOLOCK flag
+
+	* apps/app_queue.c: Remove some ref leaks and a return without
+	  unlock. There some resource leaks introduced in asterisk 10 make
+	  sure that locks are not held on return and we release ref's held.
+
+	* /, apps/app_queue.c: Revert Janitor patch 341920 For now
+
+	* /, apps/app_queue.c: Whitespace Fixups / Add Braces This
+	  janitorial patch is related to work on RB1538 ........ Merged
+	  revisions 341906 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-21 16:42 +0000 [r341807-341810]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395
+	  ........ Merged revisions 341809 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, pbx/pbx_lua.c: don't limit the length of app and function
+	  arguments ASTERISK-18395 ........ Merged revisions 341806 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 21:58 +0000 [r341718]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/features.h, /, main/features.c, res/res_agi.c:
+	  Fix AGI exec Park to honor the Park application parameters. The
+	  fix for ASTERISK-12715 and ASTERISK-12685 added a check for the
+	  Park application because the channel needed to be masqueraded to
+	  prevent a crash. Since the Park application now always
+	  masquerades the channel into the parking lot, the special check
+	  is no longer needed. The fix also resulted in AGI exec Park
+	  attempting to double park the call and not honor the Park
+	  application parameters. * Removed no longer necessary call to
+	  ast_masq_park_call() by AGI exec for the Park application.
+	  (Reverts -r146923) * Fix Park application to only return 0 or -1.
+	  The AGI exec Park was causing broken pipe error messages because
+	  the Park application returned 1 on successful park. (closes issue
+	  ASTERISK-18737) ........ Merged revisions 341717 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 21:27 +0000 [r341665-341707]  Paul Belanger <pabelanger at digium.com>
+
+	* /, funcs/func_callerid.c: Fixed typo from previous commit
+	  ........ Merged revisions 341704 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, funcs/func_callerid.c: Updated documentation for the optional
+	  CID parameter with CALLERID ........ Merged revisions 341664 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-20 18:20 +0000 [r341580-341599]  Gregory Nietsky <gregory at distrotech.co.za>
+
+	* configs/queues.conf.sample: add documentation for
+	  check_state_unknown in configs/queues.conf.sample app_queue
+	  allows calls to members in a "Unknown" state to be treated as
+	  available setting check_state_unknown = yes will cause app_queue
+	  to query the channel driver to better determine the state this
+	  only applies to queues with ringinuse or ignorebusy set
+	  appropriately.
+
+	* CHANGES, apps/app_queue.c: Add option to check state when state
+	  is unknown r341486 reverts r325483 this is a rework of the patch.
+	  optimize to minimize load. add option check_state_unknown to
+	  control whether a member with unknown device state is checked
+	  there is a small % chance that calls will be sent to the member
+	  when they on a call. app_queue will see a device with unknown
+	  state as available and does not try verify the state without this
+	  option enabled. Review: https://reviewboard.asterisk.org/r/1535/
+
+2011-10-20 15:14 +0000 [r341530]  Terry Wilson <twilson at digium.com>
+
+	* /, include/asterisk/strings.h: Clean up ast_check_digits The code
+	  was originally copied from the is_int() function in the AEL code.
+	  wdoekes pointed out that the function should take a const char*
+	  and that their was an unneeded variable. This is now fixed.
+	  ........ Merged revisions 341529 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 21:23 +0000 [r341486]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: Fix a performance regression introduced in
+	  r325483. The regression was caused by a call to
+	  ast_parse_device_state() in app_queue's ring_entry() function.
+	  The ast_parse_device_state() function eventually calls
+	  ast_channel_get_full() with a channel name prefix which causes it
+	  to walk the channel list causing massive lock contention and slow
+	  downs. This patch fixes the regression by removing the call to
+	  ast_parase_device_state() which should be unnecessary. Queue
+	  member device state should be maintained by device state events.
+	  Some users have seen instances where busy agents were called when
+	  they shouldn't have, which is the reason the call to
+	  ast_parse_device_state() was added. That change appears to have
+	  resolved that issue but also causes this performance regression.
+	  There may still be issues with queue member status, and if so,
+	  alternative methods should be investigated to resolve them.
+	  AST-695
+
+2011-10-19 19:01 +0000 [r341436]  Paul Belanger <pabelanger at digium.com>
+
+	* /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google
+	  has recently make some changes (again) to their protocol. Rather
+	  then patching asterisk to flip between the two different methods,
+	  we now allow both. Lets hope this keeps Google Voice happy for a
+	  while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
+	  Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
+	  6311) ........ Merged revisions 341435 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 07:42 +0000 [r341380]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c, include/asterisk/strings.h: Don't use
+	  is_int() since it doesn't link well on all platforms Just create
+	  an normal API function in strings.h that does the same thing just
+	  to be safe. ASTERISK-17146 ........ Merged revisions 341379 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-19 07:23 +0000 [r341377]  Stefan Schmidt <sst at sil.at>
+
+	* /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
+	  when Asterisk has not yet received a Contact URI from a UAS
+	  ........ Merged revisions 341366 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-18 23:42 +0000 [r341315]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Don't resolve numeric hosts or contact
+	  unresolved hosts If a SIP dial string contains a numeric hostname
+	  that is not a peer name, don't try to resolve it as it is
+	  unlikely that someone really means Dial(SIP/0.0.4.26) when
+	  Dial(SIP/1050) is called. Also, make sure that create_addr
+	  returns -1 if an address isn't resolved so that we don't attempt
+	  to send SIP requests to an address that doesn't resolve. (closes
+	  issue ASTERISK-17146, ASTERISK-17716) Review:
+	  https://reviewboard.asterisk.org/r/1532/ ........ Merged
+	  revisions 341314 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-18 23:33 +0000 [r341313]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c, /: Merged revisions 341312 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.8
+	  ........ r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct
+	  2011) | 3 lines fix issue on channel numbering (calls could have
+	  same channel number on heavy loaded system) ........
+
+2011-10-18 21:11 +0000 [r341255]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_mgcp.c, include/asterisk/features.h,
+	  channels/chan_dahdi.c, channels/sig_analog.c, /,
+	  channels/chan_sip.c, main/features.c, channels/chan_iax2.c,
+	  channels/sip/include/sip.h: More parking issues. * Fix potential
+	  deadlocks in SIP and IAX blind transfer to parking. * Fix SIP,
+	  IAX, DAHDI analog, and MGCP channel drivers to respect the
+	  parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
+	  parameter). Created ast_park_call_exten() and
+	  ast_masq_park_call_exten() to maintian API compatibility. * Made
+	  masq_park_call() handle a failed ast_channel_masquerade() setup.
+	  * Reduced excessive struct parkeduser.peername[] size. ........
+	  Merged revisions 341254 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 17:36 +0000 [r341190]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Initialize variables before calling
+	  parse_uri If parse_uri was called with an empty URI, some
+	  pointers would be modified and an invalid read could result. This
+	  patch avoids calling parse_uri with an empty contact uri when
+	  parsing REGISTER requests. AST-2011-012 (closes issue
+	  ASTERISK-18668) ........ Merged revisions 341189 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 16:53 +0000 [r341148]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
+	  include of asterisk/md5.h in pbx_realtime.c . A commit needed to
+	  test the commit message. Merged-From:
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8@341074
+
+2011-10-17 16:38 +0000 [r341122-341146]  Paul Belanger <pabelanger at digium.com>
+
+	* tests/test_format_api.c: Set 'core' support level for
+	  test_format_api.c
+
+	* apps/app_voicemail.c, /: Multiple revisions 341108,341112
+	  ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon,
+	  17 Oct 2011) | 2 lines Voicemail compiler flags are 'core'
+	  support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400
+	  (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged
+	  revisions 341108,341112 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-17 16:18 +0000 [r341094]  Jason Parker <jparker at digium.com>
+
+	* CHANGES: Add information about limitations of new codec support
+	  in channel drivers. (issue ASTERISK-18680)
+
+2011-10-17 15:39 +0000 [r341089]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Don't try to remove peers without IPs
+	  from peers_by_ip (closes issue ASTERISK-18696) ........ Merged
+	  revisions 341088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 21:36 +0000 [r341023]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change
+	  the internal name of the menuselect options that are used to
+	  control whether modules are embedded or not; using just the bare
+	  category name led to accidentally enabling these options when
+	  users used the wrong "--enable" operation on the menuselect
+	  command line. Now the internal option names are prefixed with
+	  "EMBED_", so they won't be the same as the name of the category
+	  containing the modules they control the embedding of. ........
+	  Merged revisions 341022 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 20:50 +0000 [r340971]  Kinsey Moore <kmoore at digium.com>
+
+	* res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
+	  340970 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) |
+	  8 lines Quiet RTCP Receiver Reports during fax transmission RTCP
+	  is now disabled for "inactive" RTP audio streams during SIP T.38
+	  sessions. The ability to disable RTCP streams in res_rtp_asterisk
+	  was missing, so this code was added to support the bug fix.
+	  (closes issue ASTERISK-18400) ........
+
+2011-10-14 18:23 +0000 [r340931]  Jonathan Rose <jrose at digium.com>
+
+	* utils/utils.xml, funcs/func_jitterbuffer.c: Some additional
+	  module documentation changes for 10 for the menuselect change.
+	  (issue ASTERISK-18268)
+
+2011-10-14 16:39 +0000 [r340879]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c, /: Avoid unnecessary WARNING message Add
+	  AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
+	  displaying a WARNING message. (closes issue ASTERISK-18610) Patch
+	  by: Kristijan_Vrban ........ Merged revisions 340878 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-14 16:18 +0000 [r340868]  Jonathan Rose <jrose at digium.com>
+
+	* funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml,
+	  /, res/res_fax.c, apps/app_celgenuserevent.c,
+	  codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c: Fixes
+	  some support level info so that it can be read by menuselect.
+	  (issue ASTERISK-18268) Review:
+	  https://reviewboard.asterisk.org/r/1525/ ........ Merged
+	  revisions 340863 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-13 22:54 +0000 [r340810]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/features.c: Fix DTMF blind transfer continuing to execute
+	  dialplan after transfer. Party A calls Party B. Party A DTMF
+	  blind transfers Party B to Party C. Party A channel continues to
+	  execute dialplan. * Fixed the return value of
+	  builtin_blindtransfer() to return the correct value after a
+	  transfer so the dialplan will not keep executing. * Removed
+	  unnecessary connected line update that did not really do
+	  anything. * Made access to GOTO_ON_BLINDXFR thread safe in
+	  check_goto_on_transfer(). * Fixed leak of xferchan for failure
+	  cases in check_goto_on_transfer(). * Updated debug messages in
+	  builtin_blindtransfer() and check_goto_on_transfer(). (closes
+	  issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
+	  ........ Merged revisions 340809 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2011-10-13 08:46 +0000 [r340770]  Gregory Nietsky <gregory at distrotech.co.za>
+
+	* channels/chan_sip.c: Only send MWI Notify on register if the
+	  registration is successful. lastmsgssent was removed from
+	  chan_sip and the old behavior of sending a mwi notify on register
+	  [except when subscribemwi is set] was restored but this must only
+	  happen when registration succeeds. leaking information for
+	  unsuccessful registrations is not secure.
+
+2011-10-13 06:59 +0000 [r340718]  Stefan Schmidt <sst at sil.at>
+
+	* channels/chan_sip.c: Merged revisions 340717 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
+	  r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011)
+	  | 3 lines storing the route-set also on a 181 response not only
+	  on 180,182 or 183. ........
+
+2011-10-13 06:56 +0000 [r340578-340716]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Initialize ast_sockaddr before calling
+	  ast_sockaddr_resolve Avoid possible jump based on unitialized
+	  value ........ Merged revisions 340715 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, res/res_config_sqlite.c: Don't skip the query field on a
+	  realtime multi query There is no documented reason to not add the
+	  query field to the varlist returned by a realtime multi query,
+	  despite the config category being set to its value. Of course,
+	  there is no documentation that the category should be set to the
+	  value either. There is lots of no documentation when it comes to

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