[svn-commits] bebuild: tag 13.0.0-beta2 r423615 - in /tags/13.0.0-beta2: ./ contrib/realtim...

SVN commits to the Digium repositories svn-commits at lists.digium.com
Fri Sep 19 16:02:16 CDT 2014


Author: bebuild
Date: Fri Sep 19 16:02:13 2014
New Revision: 423615

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423615
Log:
Importing files for 13.0.0-beta2 release.

Added:
    tags/13.0.0-beta2/.lastclean   (with props)
    tags/13.0.0-beta2/.version   (with props)
    tags/13.0.0-beta2/ChangeLog   (with props)
    tags/13.0.0-beta2/contrib/realtime/mysql/mysql_cdr.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/mysql/mysql_config.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/mysql/mysql_voicemail.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/oracle/oracle_cdr.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/oracle/oracle_config.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/oracle/oracle_voicemail.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_cdr.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_config.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/postgresql/postgresql_voicemail.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_cdr.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_config.sql   (with props)
    tags/13.0.0-beta2/contrib/realtime/sqlserver/mssql_voicemail.sql   (with props)

Added: tags/13.0.0-beta2/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0-beta2/.lastclean?view=auto&rev=423615
==============================================================================
--- tags/13.0.0-beta2/.lastclean (added)
+++ tags/13.0.0-beta2/.lastclean Fri Sep 19 16:02:13 2014
@@ -1,0 +1,1 @@
+40

Propchange: tags/13.0.0-beta2/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/13.0.0-beta2/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/13.0.0-beta2/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/13.0.0-beta2/.version
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0-beta2/.version?view=auto&rev=423615
==============================================================================
--- tags/13.0.0-beta2/.version (added)
+++ tags/13.0.0-beta2/.version Fri Sep 19 16:02:13 2014
@@ -1,0 +1,1 @@
+13.0.0-beta2

Propchange: tags/13.0.0-beta2/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/13.0.0-beta2/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/13.0.0-beta2/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/13.0.0-beta2/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/13.0.0-beta2/ChangeLog?view=auto&rev=423615
==============================================================================
--- tags/13.0.0-beta2/ChangeLog (added)
+++ tags/13.0.0-beta2/ChangeLog Fri Sep 19 16:02:13 2014
@@ -1,0 +1,17974 @@
+2014-09-19  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 13.0.0-beta2 Released.
+
+2014-09-19 19:51 +0000 [r423580]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_pjsip_notify.c: res_pjsip_notify: Fix crash on
+	  unload/load and don't say the module doesn't exist on reload.
+	  When unloading the module did not unregister the CLI commands
+	  causing a crash upon load when they were registered again. When
+	  reloading the module the return value from the config options
+	  framework was not checked to determine if an error occurred or
+	  not. This caused a message to be output saying the module did not
+	  exist when reloading if no changes were present. AST-1433 #close
+	  AST-1434 #close ........ Merged revisions 423579 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-19 17:08 +0000 [r423561]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_pjsip.c, res/res_pjsip_sdp_rtp.c:
+	  res_pjsip_sdp_rtp.c: Fix native formats containing formats that
+	  were not negotiated. Outgoing PJSIP calls can result in
+	  non-negotiated formats listed in the channel's native formats if
+	  video formats are listed in the endpoint's configuration. The
+	  resulting call could then use a non-negotiated format resulting
+	  in one way audio. * Simplified the update of session->req_caps in
+	  set_caps(). Why do something in five steps when only one is
+	  needed? AFS-162 #close Review:
+	  https://reviewboard.asterisk.org/r/4000/
+
+2014-09-19 15:18 +0000 [r423524-423530]  Jonathan Rose <jrose at digium.com>
+
+	* /, main/stasis_channels.c: Stasis_channels: Resolve unfinished
+	  Dials when doing masquerades Masquerades into channels that are
+	  in the dialing state don't end their dial and this goes against
+	  the model for things like CDRs and generating Dial end manager
+	  actions and such. ASTERISK-24237 #close Reported by: Richard
+	  Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........
+	  Merged revisions 423525 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* channels/chan_iax2.c: chan_iax2: Fix a crash when using chan_iax2
+	  jitterbuffer settings Caused by format changes in Asterisk 13
+	  ASTERISK-24265 #close Reported by: Dafi Ni Review:
+	  https://reviewboard.asterisk.org/r/3999/
+
+2014-09-19 12:45 +0000 [r423504]  Kinsey Moore <kmoore at digium.com>
+
+	* /, main/framehook.c, res/res_pjsip_t38.c,
+	  include/asterisk/framehook.h: PJSIP: Prevent T38 framehook being
+	  put on wrong channel This change gives framehooks a
+	  reverse-direction masquerade callback in addition to
+	  chan_fixup_cb similar to the callback added to datastores to
+	  handle the same situation. The new callback provides the same
+	  parameters as the fixup callback, but is called on the new
+	  channel's framehooks before moving framehooks from the old
+	  channel to the new channel. This gives the framehooks an
+	  oppurtunity to decide whether they should remain on the new
+	  channel or be removed. This new callback is used to prevent the
+	  PJSIP T.38 framehook from remaining on a masqueraded channel if
+	  the new channel is not also a PJSIP channel. This was causing a
+	  crash when a local channel was masqueraded into a PJSIP channel
+	  and the framehook was executed on the local channel since the
+	  channel's tech private data was not structured as expected.
+	  Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged
+	  revisions 423503 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 19:30 +0000 [r423482]  Sean Bright <sean at malleable.com>
+
+	* res/res_pjsip/config_auth.c, /: res_pjsip: Don't require a
+	  password when doing userpass authentication. An empty password is
+	  valid for username/password authentication so we should allow
+	  password to be empty/not supplied. Review:
+	  https://reviewboard.asterisk.org/r/3988 ........ Merged revisions
+	  423481 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 19:22 +0000 [r423478]  George Joseph <george.joseph at fairview5.com>
+
+	* main/utils.c, include/asterisk/strings.h, tests/test_strings.c,
+	  /: utils: Create ast_strsep function that ignores separators
+	  inside quotes This function acts like strsep with three
+	  exceptions... * The separator is a single character instead of a
+	  string. * Separators inside quotes are treated literally instead
+	  of like separators. * You can elect to have leading and trailing
+	  whitespace and quotes stripped from the result and have '\'
+	  sequences unescaped. Like strsep, ast_strsep maintains no
+	  internal state and you can call it recursively using different
+	  separators on the same storage. Also like strsep, for consistent
+	  results, consecutive separators are not collapsed so you may get
+	  an empty string as a valid result. Tested by: George Joseph
+	  Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged
+	  revisions 423476 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 18:31 +0000 [r423462]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_pjsip_pubsub.c: Add subscription state test events. These
+	  are needed for a set of batched notification RLS tests that are
+	  about to be committed to the testsuite. Review:
+	  https://reviewboard.asterisk.org/r/3967
+
+2014-09-18 17:11 +0000 [r423425]  Jonathan Rose <jrose at digium.com>
+
+	* /, res/res_pjsip_endpoint_identifier_ip.c:
+	  res_pjsip_endpoint_identifier_ip: Fix parsing of match value with
+	  CIDR Also fixes comma separates match lists ASTERISK-24290 #close
+	  Reported by: Ray Crumrine Review:
+	  https://reviewboard.asterisk.org/r/3995/ ........ Merged
+	  revisions 423417 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 17:09 +0000 [r423418-423423]  Richard Mudgett <rmudgett at digium.com>
+
+	* bridges/bridge_softmix.c: bridge_softmix.c: Made use
+	  ao2_replace() instead of the inline equivalent. * Clarified some
+	  read/write format comments. * Fixed a doxygen tag typo.
+
+	* main/astobj2.c, contrib/scripts/refcounter.py, /:
+	  astobj2.c/refcounter.py: Fix to deal with invalid object refs. *
+	  Make astob2 REF_DEBUG output an invalid object line when an
+	  invalid ao2 object ref/unref is attempted. This is similar to the
+	  constructor/destructor lines. * Fixed refcounter.py to handle
+	  skewed objects that have constructor/destructor states. * Made
+	  refcounter.py highlight the invalid ao2 object refs by putting
+	  them in their own section of the processed output file. * Made
+	  refcounter.py highlight unreffing an object by more than one that
+	  results in a negative ref count and the object being destroyed.
+	  The abnormally destroyed object is reported in the invalid and
+	  finalized object sections of the output. Review:
+	  https://reviewboard.asterisk.org/r/3971/ ........ Merged
+	  revisions 423349 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 423400 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423416 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 16:37 +0000 [r423348-423414]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/format_cap.h, main/channel.c, main/format_cap.c,
+	  main/translate.c: Add API call to determine if format capability
+	  structure is "empty". Empty here means that there are no formats
+	  in the format_cap structure or the only format in it is the
+	  "none" format. I've added calls to check the emptiness of a
+	  format_cap in a few places in order to short-circuit operations
+	  that would otherwise be pointless as well as to prevent some
+	  assertions from being triggered in cases where channels with no
+	  formats are used.
+
+	* /, res/res_fax_spandsp.c: res_fax_spandsp: Properly handle
+	  cleanup before starting FAXes. If faxing fails at a very early
+	  stage, then it is possible for us to pass a NULL t30 state
+	  pointer to spandsp, which spandsp is none too pleased with. This
+	  patch ensures that we pass the correct pointer to spandsp in the
+	  situation where we have not yet set our local t30 state pointer.
+	  ASTERISK-24301 #close Reported by Matt Jordan Patches:
+	  ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License
+	  #5049) ........ Merged revisions 423360 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423365 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* /, res/res_pjsip_mwi.c,
+	  res/res_pjsip_dialog_info_body_generator.c,
+	  res/res_pjsip_xpidf_body_generator.c,
+	  res/res_pjsip_mwi_body_generator.c, res/res_pjsip_pubsub.c,
+	  res/res_pjsip_exten_state.c, include/asterisk/res_pjsip_pubsub.h,
+	  res/res_pjsip_pidf_body_generator.c: res_pjsip_pubsub: Add some
+	  type safety when generating NOTIFY bodies. res_pjsip_pubsub has
+	  two separate checks that it makes when a SUBSCRIBE arrives. * It
+	  checks that there is a subscription handler for the Event * It
+	  checks that there are body generators for the types in the Accept
+	  header The problem is, there's nothing that ensures that these
+	  two things will actually mesh with each other. For instance,
+	  Asterisk will accept a subscription to MWI that accepts pidf+xml
+	  bodies. That doesn't make sense. With this commit, we add some
+	  type information to the mix. Subscription handlers state they
+	  generate data of type X, and body generators state that they
+	  consume data of type X. This way, Asterisk doesn't end up in some
+	  hilariously mismatched situation like the one in the previous
+	  paragraph. ASTERISK-24136 #close Reported by Mark Michelson
+	  Review: https://reviewboard.asterisk.org/r/3877 Review:
+	  https://reviewboard.asterisk.org/r/3878 ........ Merged revisions
+	  423344 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 15:13 +0000 [r423284]  George Joseph <george.joseph at fairview5.com>
+
+	* res/res_pjsip_endpoint_identifier_ip.c,
+	  res/res_pjsip/pjsip_configuration.c,
+	  res/res_pjsip/pjsip_options.c, res/res_pjsip/config_transport.c,
+	  include/asterisk/res_pjsip.h, res/res_pjsip/config_auth.c, /,
+	  res/res_pjsip/location.c: res_pjsip: ami: Fix error in AMI output
+	  when an endpoint has no transport When no transport is associated
+	  to an endpoint, the AMI output for PJSIPShowEndpoint indicates an
+	  error instead of silently ignoring the missing transport. This
+	  patch causes the error to appear only if a transport was
+	  specified on the endpoint and the transport doesn't exist. It
+	  also fixes an issue with counting the objects that were actually
+	  found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by:
+	  George Joseph Review: https://reviewboard.asterisk.org/r/3998/
+	  ........ Merged revisions 423282 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-18 15:00 +0000 [r423281]  David M. Lee <dlee at digium.com>
+
+	* Makefile, makeopts.in: Only install dahdi_span_config_hook if
+	  DAHDI is enabled This patch changes the install to only install
+	  the hook script if DAHDI is enabled. It also adds the script to
+	  the uninstall task, and moves the DAHDI_UDEV_HOOK_DIR variable so
+	  that it's not between the _MAKEOPTS variables and their comment.
+	  This allows installs which specify a --prefix to work normally,
+	  as long as they don't enable DAHDI. Review:
+	  https://reviewboard.asterisk.org/r/3972/
+
+2014-09-18 14:45 +0000 [r423279]  George Joseph <george.joseph at fairview5.com>
+
+	* include/asterisk/config.h, main/config.c, main/manager.c, /:
+	  config: bug: Fix SEGV in ast_category_insert when matching
+	  category isn't found If you call ast_category_insert with a match
+	  category that doesn't exist, the list traverse runs out of 'next'
+	  categories and you get a SEGV. This patch adds check for the
+	  end-of-list condition and changes the signature to return an int
+	  for success/failure indication instead of a void. The only
+	  consumer of this function is manager and it was also changed to
+	  use the return value. Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3993/ ........ Merged
+	  revisions 423276 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 423277 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423278 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-17 18:05 +0000 [r423209-423255]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_rtp_asterisk.c: res_rtp_asterisk: Ensure that the
+	  thread terminating pj stuff is registered. ........ Merged
+	  revisions 423253 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423254 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix 100% CPU usage
+	  due to timer heap thread spinning. Side note: I need a vacation.
+	  ........ Merged revisions 423210 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423211 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* res/res_rtp_asterisk.c, /: res_rtp_asterisk: Fix building when
+	  pjproject is not used. ........ Merged revisions 423207 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423208 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-16 16:32 +0000 [r423192]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/file.c, apps/app_voicemail.c, include/asterisk/file.h:
+	  Voicemail: get correct duration when copying file to vm Changes
+	  made during format improvements resulted in the recording to
+	  voicemail option 'm' of the MixMonitor app writing a zero length
+	  duration in the msgXXXX.txt file. This change introduces a new
+	  function ast_ratestream(), which provides the sample rate of the
+	  format associated with the stream, and updates the app_voicemail
+	  function for ast_app_copy_recording_to_vm to calculate the right
+	  duration. Review: https://reviewboard.asterisk.org/r/3996/
+	  ASTERISK-24328 #close
+
+2014-09-16 12:12 +0000 [r423152-423173]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_session.c, /: res_pjsip_session: Fix usage of wrong
+	  memory pool when creating local SDP. ........ Merged revisions
+	  423172 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* include/asterisk/rtp_engine.h, res/res_rtp_asterisk.c, /:
+	  res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The
+	  number of file descriptors an ioqueue instance can handle is
+	  fixed, so we now spawn the required number to handle the load. 2.
+	  Our transport identifiers were exceeding the range supported by
+	  pjnath. 3. The TURN client did not set up client binding causing
+	  needless bandwidth usage. 4. The code no longer updates address
+	  information on each packet. 5. STUN traffic was getting looped
+	  back to Asterisk instead of going through the TURN server. 6.
+	  Synchronization now ensures things are completely setup or
+	  destroyed. 7. Logging now reflects the target the TURN server is
+	  sending to/receiving from on our behalf. ASTERISK-23577 #close
+	  Reported by: Jay Jideliov ASTERISK-23634 #close Reported by:
+	  Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/
+	  ........ Merged revisions 423150 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423151 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-15 10:49 +0000 [r423069-423129]  Walter Doekes <walter+asterisk at wjd.nu>
+
+	* /,
+	  contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py
+	  (added): contrib: Fix verifyi typo in alembic DB script
+	  ps_transport table. Reported by: Zogot (on IRC) Patches: tmp.diff
+	  uploaded by Zogot, cleaned up by me. ........ Merged revisions
+	  423128 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* /, configs/samples/sip.conf.sample: chan_sip: Clarify that
+	  sipdebug=yes cannot be undone by the CLI. Document it in
+	  sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod
+	  Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged
+	  revisions 423066 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 423067 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 423068 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-12 16:09 +0000 [r422985]  Jonathan Rose <jrose at digium.com>
+
+	* main/config.c, /: Realtime: Fix a bug that caused realtime
+	  destroy command to crash Also has could affect with anything that
+	  goes through ast_destroy_realtime. If a CLI user used the command
+	  'realtime destroy <family>' with only a single column/value pair,
+	  Asterisk would crash when trying to create a variable list from a
+	  NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson
+	  Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged
+	  revisions 422984 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-11 22:16 +0000 [r422965]  Mark Michelson <mmichelson at digium.com>
+
+	* /, main/app.c: Remove undocumented default behavior of
+	  ast_play_and_record_full acceptdtmf. ast_play_and_record_full()
+	  has a parameter called "acceptdtmf" that is a string of
+	  acceptable DTMF digits that may be pressed by a caller to end and
+	  accept the recording. ARI uses this function in order to perform
+	  recording, and it provides options for what is passed as
+	  acceptdtmf to ast_play_and_record_full(). By default, ARI passes
+	  an empty string, with the intention that no DTMF can be used to
+	  end the recording. The problem is that ast_play_and_record_full()
+	  attempts to be "helpful" by setting "#" as the acceptdtmf if an
+	  empty string or NULL pointer has been passed in. With ARI, this
+	  results in unexpected behavior occurring if you have attempted to
+	  intercept "#" yourself in order to perform some other
+	  manipulation of the live recording. This change removes the
+	  "helpful" behavior by no longer accepting "#" as a default
+	  acceptdtmf if none is specified by the caller of
+	  ast_play_and_record_full(). This makes the ARI scenario work as
+	  expected. The other callers of ast_play_and_record_full() are
+	  app_voicemail and app_minivm, and in both cases, they pass an
+	  explicit "#" to ast_play_and_record_full() as acceptdtmf, so they
+	  are unaffected by this change. ........ Merged revisions 422964
+	  from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-10 16:04 +0000 [r422905]  George Joseph <george.joseph at fairview5.com>
+
+	* main/config.c, /: config: bug: fix truncation of included config
+	  files on permissions error ast_config_text_file_save() currently
+	  truncates include files as they are processed. If a subsequent
+	  include file or the main config file has a permissions error that
+	  prevents writing, earlier include files are left truncated
+	  resulting in a frantic search for backups. This patch causes
+	  ast_config_text_file_save to check for write access on all files
+	  before it truncates any of them. Will be applied 1.8 > trunk.
+	  Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3986/ ........ Merged
+	  revisions 422900 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 422903 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 422904 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-10 15:59 +0000 [r422901]  Sean Bright <sean at malleable.com>
+
+	* res/res_pjsip/config_auth.c, /: pjsip/config_auth.c: Add missing
+	  whitespace to log messages. The errors generated when validating
+	  'auth' settings are missing a space which makes the messages a
+	  little confusing. ........ Merged revisions 422899 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-09 20:01 +0000 [r422883]  Rusty Newton <rnewton at digium.com>
+
+	* /, sounds/sounds.xml, sounds/Makefile: Sounds/BuildSystem:
+	  Modifications to include new releases and Japanese language.
+	  Modifying Makefile and sounds.xml to include new core 1.4.26 and
+	  extra 1.4.15 sound prompt releases, plus the new Japanese core
+	  sound prompts contributed by QLOOG. ASTERISK-23324 Reported by:
+	  Kevin McCoy Tested by: Rusty Newton ........ Merged revisions
+	  422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+	  ........ Merged revisions 422790 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 422791 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-08 18:03 +0000 [r422851-422855]  Mark Michelson <mmichelson at digium.com>
+
+	* configs/samples/pjsip.conf.sample: Add note about configuring
+	  list_items on a single line.
+
+	* configs/samples/pjsip.conf.sample: Add sample configuration for
+	  resource lists. On review /r/3977, it was recommended to note in
+	  the sample configuration about the size limitation for resource
+	  lists. However, since there was no section in the sample
+	  configuration at all for resource list subscriptions, I decided
+	  to make a separate commit where I have added the necessary sample
+	  configuration as well as the size limitation warning.
+
+	* res/res_pjsip_pubsub.c: Pre-allocate transmission data buffer for
+	  RLS NOTIFY requests. PJSIP, unless a constant is modified at
+	  compilation time, limits SIP requests to 4000 bytes. Full-state
+	  RLS notifications can easily exceed this limit with moderately
+	  small lists. This changeset allows for Asterisk to work around
+	  this size limit by performing its own allocation of the
+	  transmission data buffer. This way, Asterisk can allocate a
+	  buffer that exceeds the built-in maximum. We still impose our own
+	  limit of 64000 bytes, mainly because making allocations larger
+	  than that is a bit absurd. ASTERISK-24181 #close Reported by Mark
+	  Michelson Review: https://reviewboard.asterisk.org/r/3977
+
+2014-09-08 15:41 +0000 [r422836]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_pjsip_pubsub.c: res_pjsip_pubsub: Check supported headers
+	  for eventlist when subscribing to resource list
+	  https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
+	  According to the off-nominal plan, if evenlist support is not
+	  specified in a SUBSCRIBE's supported header(s), that subscription
+	  should be rejected with an error. ASTERISK-23871 Reported by:
+	  Mark Michelson Review:
+	  https://reviewboard.asterisk.org/r/3960/diff/#index_header
+
+2014-09-06 22:49 +0000 [r422767-422770]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cdr.c, /: main/cdr: Copy over location information during a
+	  fork When a CDR is forked, a new CDR is created and appended to
+	  the CDR chain for the Party A. The forked CDR starts life off as
+	  a clone of the last non-finalized for the particular Party A. In
+	  the past, merely copying over the snapshots for Party A/Party B
+	  would be sufficient. However, as the CDRs now contain cached
+	  information from Party A - specifically application/data,
+	  context, and extension - we need to copy that over during a fork
+	  as well. Huzzah for unit tests catching this when the
+	  context/extension were derived from a cached value on the CDR
+	  instead of on Party A. ........ Merged revisions 422769 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* main/rtp_engine.c, /: main/rtp_engine: Format NTP timestamps as
+	  unsigned ints On some systems, a timeval's tv_sec/tv_usec will be
+	  unsigned lont ints, as opposed to long ints. When the RTP engine
+	  formats these as strings, it was previously formatting them as
+	  signed integers, which can result in some odd negative timestamp
+	  values (particularly on 32-bit systems). This patch formats the
+	  values as unsigned long integers. ........ Merged revisions
+	  422766 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-06 19:12 +0000 [r422747]  Joshua Colp <jcolp at digium.com>
+
+	* /, res/res_pjsip_sdp_rtp.c: res_pjsip_sdp_rtp: Fix retrieval of
+	  "ice-pwd" attribute if in session and not media stream. ........
+	  Merged revisions 422746 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 22:03 +0000 [r422716-422719]  Matthew Jordan <mjordan at digium.com>
+
+	* /, apps/app_macro.c, include/asterisk/channel.h,
+	  apps/app_stack.c, main/cdr.c: main/cdrs: Preserve
+	  context/extension when executing a Macro or GoSub The
+	  context/extension in a CDR is generally considered the
+	  destination of a call. When looking at a 2-party call CDR, users
+	  will typically be presented with the following: context exten
+	  channel dest_channel app data default 1000 SIP/8675309 SIP/1000
+	  Dial SIP/1000,,20 However, if the Dial actually takes place in a
+	  Macro, the current behaviour in 12 will result in the following
+	  CDR: context exten channel dest_channel app data macro-dial s
+	  SIP/8675309 SIP/1000 Dial SIP/1000,,20 The same is true of a
+	  GoSub: context exten channel dest_channel app data subs
+	  dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This generally
+	  makes the context/exten fields less than useful. It isn't hard to
+	  preserve these values in the CDR state machine; however, we need
+	  to have something that informs us when a channel is executing a
+	  subroutine. Prior to this patch, there isn't anything that does
+	  this. This patch solves this problem by adding a new channel
+	  flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel
+	  when it executes a Macro or a GoSub. The CDR engine looks for
+	  this value when updating a Party A snapshot; if the flag is
+	  present, we don't override the context/exten on the main CDR
+	  object. In a funny quirk, executing a hangup handler must *not*
+	  abide by this logic, as the endbeforehexten logic assumes that
+	  the user wants to see data that occurs in hangup logic, which
+	  includes those subroutines. Since those execute outside of a
+	  typical Dial operation (and will typically have their own
+	  dedicated CDR anyway), this is unlikely to cause any heartburn.
+	  Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254
+	  #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis
+	  ........ Merged revisions 422718 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* main/cdr.c, /: main/cdr: Fix crash/memory consumption in CDRs in
+	  multi-party bridge scenarios This patch fixes an issue where CDRs
+	  would get stuck generating an infinite number of CDRs, eventually
+	  crashing Asterisk (and consuming a lot of memory along the way).
+	  When a channel enters into a multi-party bridge, the CDR engine
+	  creates mappings of each participant to each other participant,
+	  picking the 'A' party as it goes. So, if we have four channels in
+	  a multi-party bridge (Alice, Bob, Charlie, Denise), we would have
+	  something like: Alice => Bob Alice => Charlie Alice => Denise Bob
+	  => Charlie Bob => Denise Charlie => Denise This works fine when
+	  participants enter the bridge a single time. When a participant
+	  leaves a bridge, the CDRs for that channel are transitioned to a
+	  finalized state. The bug occurs if Bob rejoins. When the CDR
+	  engine creates mappings between the channels, it walks through
+	  all the participants currently in the bridge, and realizes that
+	  no one in the bridge can create a CDR with the channel (Bob). As
+	  such it creates a new CDR for the candidate and appends it to
+	  that candidate's chain. Unfortunately, on this particular code
+	  path, it doesn't stop traversing the candidate's chain. Since we
+	  just added ourselves to the chain, this causes the loop to keep
+	  going, constantly adding new CDRs. This patch makes it so the
+	  engine bails when it creates a CDR match in this case. Review:
+	  https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close
+	  Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat
+	  ASTERISK-24208 Reported by: Frankie Chin ........ Merged
+	  revisions 422715 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 20:35 +0000 [r422700]  Richard Mudgett <rmudgett at digium.com>
+
+	* funcs/func_channel.c: func_channel.c: Add missing locking to some
+	  CHANNEL() requests. * The CHANNEL() audionativeformat,
+	  videonativeformat, audioreadformat, and audiowriteformat now need
+	  locking since the media format rework when accessing the
+	  channel's format pointers. * Increased the buffer size for
+	  CHANNEL() audionativeformat and videonativeformat output strings
+	  since the allow=all can be a lengthy list. * Tweaked the
+	  CHANNEL() XML documentation for secure_bridge_signaling,
+	  secure_bridge_media, and state. * Ensured the output buffer is
+	  initialized for secure_bridge_signaling and secure_bridge_media.
+	  * Made use the locked_copy_string() macro instead of inlining it
+	  for trace and checkhangup.
+
+2014-09-05 20:11 +0000 [r422665-422684]  Jonathan Rose <jrose at digium.com>
+
+	* include/asterisk/dial.h, main/dial.c: Dial API: Add a dial option
+	  to indicate the dialed channel will replace dialer Adds an option
+	  to the dial API that marks an outgoing dial as replacing the
+	  dialing channel for the purpose of propagating accountcode. When
+	  it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of
+	  AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on
+	  the involved channels with ast_channel_req_accountcodes. Review:
+	  https://reviewboard.asterisk.org/r/3968/
+
+	* main/cli.c, /: Call IDs: Fix appearance of call ID in core show
+	  channels when NULL NULL call IDs were meant to appear as '(none)'
+	  but instead were showing the contents of an uninitialized
+	  character buffer. ASTERISK-24223 Review:
+	  https://reviewboard.asterisk.org/r/3979/ ........ Merged
+	  revisions 422664 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-05 17:36 +0000 [r422661]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/devicestate.c, channels/chan_iax2.c: devicestate.c: Minor
+	  tweaks * In ast_state_chan2dev() use ARRAY_LEN() instead of a
+	  sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c.
+
+2014-09-05 13:28 +0000 [r422646]  Kinsey Moore <kmoore at digium.com>
+
+	* menuselect/menuselect.c: Menuselect: Fix incorrect enabling on
+	  failed deps This corrects a situation where menuselect can
+	  incorrectly enable a module by default that has defaultenabled
+	  set to "no" and has failed/non-selected dependencies. The bug is
+	  due to an inverted test when checking for whether the given
+	  module should be set to enabled by default on load. Review:
+	  https://reviewboard.asterisk.org/r/3975/ Reported by: John
+	  Bigelow
+
+2014-09-04 21:23 +0000 [r422631]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c, /: Manager: Require read permission for SYSTEM in
+	  order to send FullyBooted Review:
+	  https://reviewboard.asterisk.org/r/3969/ ........ Merged
+	  revisions 422584 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
+	  revisions 422625 from
+	  http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
+	  revisions 422626 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-03 14:05 +0000 [r422558]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_pjsip_transport_websocket.c, /:
+	  res_pjsip_transport_websocket: Fix crash when the Contact header
+	  is not a URI. The code for changing the Contact header wrongly
+	  assumed that the Contact would always contain a URI. This is
+	  incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged
+	  revisions 422557 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-02 20:29 +0000 [r422542]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_pjsip.c, res/res_pjsip_diversion.c,
+	  res/res_pjsip_session.c, include/asterisk/res_pjsip_session.h, /:
+	  Resolve race condition where channels enter dialplan application
+	  before media has been negotiated. Testsuite tests will
+	  occasionally fail because on reception of a 200 OK SIP response,
+	  an AST_CONTROL_ANSWER frame is queued prior to when media has
+	  finished being negotiated. This is because session supplements
+	  are called into before PJSIP's inv_session code has told us that
+	  media has been updated. Sometimes the queued answer frame is
+	  handled by the PBX thread before the ensuing media negotiations
+	  occur, causing a test failure. As it turns out, there is another
+	  place that session supplements could be called into, which is
+	  after media has finished getting negotiated. What this commit
+	  introduces is a means for session supplements to indicate when
+	  they wish to be called into when handling an incoming SIP
+	  response. By default, all session supplements will be run at the
+	  same point that they were prior to this commit. However, session
+	  supplements may indicate that they wish to be handled earlier
+	  than normal on redirects, or they may indicate they wish to be
+	  handled after media has been negotiated. In this changeset, two
+	  session supplements have been updated to indicate a preference
+	  for when they should be run: res_pjsip_diversion executes before
+	  handling redirection in order to get information from the
+	  Diversion header, and chan_pjsip now handles responses to INVITEs
+	  after media negotiation to fix the race condition mentioned
+	  previously. ASTERISK-24212 #close Reported by Matt Jordan Review:
+	  https://reviewboard.asterisk.org/r/3930 ........ Merged revisions
+	  422536 from http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-09-01 14:16 +0000 [r422504-422507]  Matthew Jordan <mjordan at digium.com>
+
+	* main/cli.c, /: main/cli: Do not attempt to show CDR data for
+	  internal channels Internal channels don't have CDRs. Querying the
+	  CDR engine for their variables will make it cranky. ........
+	  Merged revisions 422506 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* res/stasis/stasis_bridge.c, res/res_stasis.c, /: res_stasis:
+	  Don't play MoH to channels by default when added to holding
+	  bridges When ARI manipulates a bridge, it generally doesn't care
+	  what the mixing technology is. Operations on a bridge initiated
+	  through ARI should perform their action in generally the same
+	  way, regardless of the bridge's mixing technology. While the
+	  mixing technology may determine how media flows to channels, the
+	  actual operations on a bridge themselves should be the same.
+	  Currently, this isn't the case with holding bridges. When a
+	  channel joins without a role, MoH is started on that channel
+	  automatically. Subsequent bridge operations that would stop MoH
+	  would fail (as there is no Announcer channel playing MoH to the
+	  bridge). Starting MoH on the bridge will also create two MoH
+	  streams: one from the MoH being played on the participant
+	  channel, and one from the announcer channel. From the perspective
+	  of ARI users, this is counter-intuitive - I would not expect MoH
+	  to be started for me. The mixing technology determines how media
+	  is shared between participants, not the application experience.
+	  This patch does the following: * The Stasis bridge class now
+	  inspects channels as they are going into a bridge. If the bridge
+	  has a holding capability, and the channel has no roles, we give
+	  it a participant role and mark the default behaviour to have no
+	  entertainment. This allows addChannel operations to continue to
+	  set a participant role with an entertainment option if it felt
+	  like it (or could do it). * The music on hold channel is now
+	  Stasis approved (tm) Review:
+	  https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close
+	  Reported by: Samuel Galarneau Tested by: Samuel Galarneau
+	  ........ Merged revisions 422503 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+2014-08-30 17:32 +0000 [r422442-422445]  George Joseph <george.joseph at fairview5.com>
+
+	* /, apps/app_confbridge.c: confbridge: Add Duration to
+	  ConfbridgeList event The ConfbridgeList event doesn't include how
+	  long the user has been a member of the conference. This patch
+	  adds Duration (seconds) which is based on user->chan->answertime.
+	  Tested by: George Joseph Review:
+	  https://reviewboard.asterisk.org/r/3955/ ........ Merged
+	  revisions 422444 from
+	  http://svn.asterisk.org/svn/asterisk/branches/12
+
+	* main/manager.c, /: manager: Make WaitEvent action respect

[... 21220 lines stripped ...]



More information about the svn-commits mailing list