[asterisk-commits] bebuild: tag 11.9.0-rc1 r411553 - /tags/11.9.0-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Mar 28 13:29:55 CDT 2014


Author: bebuild
Date: Fri Mar 28 13:29:47 2014
New Revision: 411553

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=411553
Log:
Importing files for 11.9.0-rc1 release.

Added:
    tags/11.9.0-rc1/.lastclean   (with props)
    tags/11.9.0-rc1/.version   (with props)
    tags/11.9.0-rc1/ChangeLog   (with props)

Added: tags/11.9.0-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/11.9.0-rc1/.lastclean?view=auto&rev=411553
==============================================================================
--- tags/11.9.0-rc1/.lastclean (added)
+++ tags/11.9.0-rc1/.lastclean Fri Mar 28 13:29:47 2014
@@ -1,0 +1,1 @@
+40

Propchange: tags/11.9.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.9.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.9.0-rc1/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.9.0-rc1/.version
URL: http://svnview.digium.com/svn/asterisk/tags/11.9.0-rc1/.version?view=auto&rev=411553
==============================================================================
--- tags/11.9.0-rc1/.version (added)
+++ tags/11.9.0-rc1/.version Fri Mar 28 13:29:47 2014
@@ -1,0 +1,1 @@
+11.9.0-rc1

Propchange: tags/11.9.0-rc1/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/11.9.0-rc1/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/11.9.0-rc1/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/11.9.0-rc1/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/11.9.0-rc1/ChangeLog?view=auto&rev=411553
==============================================================================
--- tags/11.9.0-rc1/ChangeLog (added)
+++ tags/11.9.0-rc1/ChangeLog Fri Mar 28 13:29:47 2014
@@ -1,0 +1,29551 @@
+2014-03-28  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 11.9.0-rc1 Released.
+
+2014-03-28 17:44 +0000 [r411531]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/oochannels.c,
+	  addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c,
+	  addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c,
+	  addons/chan_ooh323.c: process stack command even if gatekeeper
+	  client isn't register don't destroy gatekeeper client if it is
+	  not started don't destroy gatekeeper client in some sort of
+	  gatekeeper errors signal rtp create condition when call cleared
+	  before rtp structure created (closes issue ASTERISK-23460)
+	  Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch
+	  Tested by: Dmitry Melekhov
+
+2014-03-28 16:16 +0000 [r411463]  Scott Griepentrog <sgriepentrog at digium.com>
+
+	* main/manager.c, /, main/http.c, main/tcptls.c: http: response
+	  body often missing after specific request This patch works around
+	  a problem with the HTTP body being dropped from the response to a
+	  specific client and under specific circumstances: a) Client
+	  request comes from node.js user agent "Shred" via use of
+	  swagger-client library. b) Asterisk and Client are *not* on the
+	  same host or TCP/IP stack In testing this problem, it has been
+	  determined that the write of the HTTP body is lost, even if the
+	  data is written using low level write function. The only solution
+	  found is to instruct the TCP stack with the shutdown function to
+	  flush the last write and finish the transmission. See review for
+	  more details. ASTERISK-23548 #close (closes issue ASTERISK-23548)
+	  Reported by: Sam Galarneau Review:
+	  https://reviewboard.asterisk.org/r/3402/ ........ Merged
+	  revisions 411462 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-28 15:43 +0000 [r411373-411458]  Matthew Jordan <mjordan at digium.com>
+
+	* /, UPGRADE.txt: UPGRADE: Note IAX2 compatibility issue between
+	  1.4 and 1.8+ systems. ........ Merged revisions 411457 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* configs/res_odbc.conf.sample, include/asterisk/res_odbc.h,
+	  res/res_config_odbc.c, /, res/res_odbc.exports.in, UPGRADE.txt,
+	  res/res_odbc.c: res_config_odbc/res_odbc: Fix handling of
+	  non-text columns updates with empty values. This patch fixes
+	  setting nullable integer columns to NULL instead of an empty
+	  string, which fails for PostgreSQL, for example. The current code
+	  is supposed to do so, but the check is broken. The patch also
+	  allows the first column in the list to be a nullable integer.
+	  This patch also adds a compatibility setting in res_odbc.conf,
+	  allow_empty_string_in_nontext. It is enabled by default. It
+	  should be disabled for database backends (such as PostgreSQL)
+	  that require NULL instead of an empty string for Integer columns.
+	  Review: https://reviewboard.asterisk.org/r/3375 (issue
+	  ASTERISK-23459) Reported by: zvision patches:
+	  res_config_odbc.diff uploaded by zvision (License 5755) ........
+	  Merged revisions 411399 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* channels/sip/include/sip.h, /: chan_sip: Add MESSAGE request to
+	  allowed methods The allowed methods advertised by chan_sip did
+	  not previously note the MESSAGE request. Even in Asterisk 1.8, we
+	  do accept in-dialog MESSAGE requests; we should advertise that we
+	  support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504
+	  #comment Reported by: Martin Kontsek ASTERISK-23504 #comment
+	  Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587)
+	  Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged
+	  revisions 411372 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-27 19:13 +0000 [r411310-411314]  Corey Farrell <git at cfware.com>
+
+	* funcs/func_strings.c, funcs/func_math.c,
+	  funcs/func_jitterbuffer.c, res/res_xmpp.c, channels/chan_iax2.c,
+	  res/res_jabber.c, res/res_mutestream.c, funcs/func_global.c,
+	  apps/app_speech_utils.c, apps/confbridge/conf_config_parser.c,
+	  funcs/func_pitchshift.c, funcs/func_callcompletion.c,
+	  funcs/func_groupcount.c, funcs/func_volume.c, funcs/func_odbc.c,
+	  funcs/func_blacklist.c, funcs/func_channel.c,
+	  funcs/func_frame_trace.c, main/features.c, funcs/func_callerid.c,
+	  apps/app_stack.c, main/message.c, res/res_calendar.c,
+	  apps/app_jack.c, apps/app_voicemail.c, funcs/func_speex.c,
+	  funcs/func_dialplan.c, channels/chan_sip.c, /: Fix dialplan
+	  function NULL channel safety issues (closes issue ASTERISK-23391)
+	  Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3386/ ........ Merged
+	  revisions 411313 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* main/format.c, include/asterisk.h, main/asterisk.c: main/formats:
+	  Fix crash in ast_format_cmp during non-clean shutdown. * Backport
+	  ast_register_cleanup from Asterisk 12. * Use ast_register_cleanup
+	  for format_attr_shutdown. ast_register_cleanup was originally
+	  commited in r390122 by dlee. (closes issue ASTERISK-23103)
+	  Reported by: JoshE
+
+2014-03-26 22:44 +0000 [r411244]  Joshua Colp <jcolp at digium.com>
+
+	* /, main/say.c: say: Fix a bug where SayNumber in Polish tries to
+	  play incorrect sound. This change fixes a bug where calling
+	  SayNumber with a number divisible by 100 using the Polish
+	  language would cause the code to attempt to play a sound file
+	  with an empty name. (closes issue ASTERISK-23509) Reported by:
+	  zvision Review: https://reviewboard.asterisk.org/r/3378/ ........
+	  Merged revisions 411243 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-26 15:57 +0000 [r411190]  Jonathan Rose <jrose at digium.com>
+
+	* configs/sip.conf.sample, /, channels/chan_sip.c: chan_sip: Send
+	  real CallerID information with P-Assserted-Identity (RFC-3325)
+	  Prior too this patch, the P-Asserted-Identity header would
+	  include anonymous caller id information which seems to go against
+	  the point of the P-Asserted-Identity header. Now the real caller
+	  ID information will be included in this header. Also, no privacy
+	  header would be included. This patch adds 'Privacy: id' to
+	  outgoing SIP messages that include the P-Asserted-Identity
+	  header. (closes issue AST-1301) ........ Merged revisions 411189
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-25 15:52 +0000 [r411089]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Fix incorrect use of timers If
+	  update_provisional_keepalive() is called while
+	  send_provisional_keepalive_full() is waiting on the PVT lock,
+	  then pvt->provisional_keepalive_sched_id will be changed to a new
+	  sched_id value by update_provisional_keepalive(), but that new
+	  sched_id then may be overwritten with -1 by
+	  send_provisional_keepalive_full(), killing the pvt's reference to
+	  a schedule and "leaking" the reference. (closes issue
+	  ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/
+	  Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies
+	  Patches: provisional_keepalive_fix.diff uploaded by Steve Davies
+	  (license 5012) ........ Merged revisions 411088 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-24 21:37 +0000 [r411022]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Always use fromdomain if set
+	  for domain, even if callerid is set to restricted. (closes issue
+	  ASTERISK-20841) Reported by: Kelly Goedert ........ Merged
+	  revisions 411021 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-20 22:46 +0000 [r410965]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_confbridge.c: app_confbridge: Fix bug - users with
+	  startmuted set don't start muted (closes issue ASTERISK-23461)
+	  Reported by: Chico Manobela Review:
+	  https://reviewboard.asterisk.org/r/3373/
+
+2014-03-18 11:50 +0000 [r410829]  Sean Bright <sean at malleable.com>
+
+	* res/res_fax_spandsp.c: res_fax_spandsp: Use g711_free() when
+	  available. Per Johann Steinwendtner on the asterisk-dev mailing
+	  list:
+	  http://lists.digium.com/pipermail/asterisk-dev/2014-March/066102.html
+	  g711_free() was introduced in spandsp 0.0.6pre4 and
+	  g711_release() became a noop. I opted not to remove the call to
+	  g711_release() since it is harmless and to call g711_free() if we
+	  have a sufficiently recent version of spandsp. (issue
+	  ASTERISK-20149) Reported by: Alexandr Gordeev
+
+2014-03-17 21:55 +0000 [r410717-410749]  Russ Meyerriecks <rmeyerreicks at digium.com>
+
+	* /, main/callerid.c: !fixup: callerid: Logic error in checksum
+	  processing Fixes syntax error in previous commit :-( ........
+	  Merged revisions 410748 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, main/callerid.c: callerid: Logic error in checksum processing
+	  Callerid checksum-ing was being handled incorrectly here. When
+	  the checksum is calculated to be 0x00, it will perform 0x100-0x00
+	  which results in 0x100. This value will then fail the otherwise
+	  correct callerid message. This patch changes the logic to simply
+	  add the calculated checksum to the transmitted 2's compliment
+	  checksum. Review: https://reviewboard.asterisk.org/r/3356/
+	  (closes issue ASTERISK-23488) ........ Merged revisions 410710
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-14 21:12 +0000 [r410609]  Jonathan Rose <jrose at digium.com>
+
+	* main/manager.c: manager: fix memory leak in manager_add_filter
+	  function (closes issue ASTERISK-23420) Reported by: Etienne
+	  Lessard Patches: manager_eventfilter_leak uploaded by Etienne
+	  Lessard (license 6394)
+
+2014-03-14 20:53 +0000 [r410556-410606]  Mark Michelson <mmichelson at digium.com>
+
+	* main/db.c: Remove an extra ast_cond_wait() that slipped through
+	  the patch.
+
+	* main/db.c: Prevent delayed astdb syncs. The syncing thread sleeps
+	  for a second before waiting to be told to attempt to sync again.
+	  If a signal were sent during this sleeping period, we would end
+	  up having to wait until the next sync signal occurred in order to
+	  sync up the astdb. This code rearrangement also ensures that any
+	  pending transactions will be synced prior to Asterisk shutting
+	  down. Patches: db_sync.patch by John Hardin (License #6512)
+
+2014-03-12 18:35 +0000 [r410490]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/confbridge/conf_state.c,
+	  apps/confbridge/conf_state_single.c,
+	  apps/confbridge/conf_state_inactive.c,
+	  apps/confbridge/conf_state_single_marked.c: app_confbridge: Make
+	  explicitly stop MOH if a user is kicked or hangs up while MOH is
+	  playing. When MOH is playing to a user in a conference and the
+	  user is kicked or hangs up from the conference then the AMI
+	  MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event:
+	  MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported
+	  by: Benjamin Keith Ford Review:
+	  https://reviewboard.asterisk.org/r/3306/
+
+2014-03-10 17:09 +0000 [r410381]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, main/http.c: AST-2014-001: Stack overflow in HTTP processing
+	  of Cookie headers. Sending a HTTP request that is handled by
+	  Asterisk with a large number of Cookie headers could overflow the
+	  stack. Another vulnerability along similar lines is any HTTP
+	  request with a ridiculous number of headers in the request could
+	  exhaust system memory. (closes issue ASTERISK-23340) Reported by:
+	  Lucas Molas, researcher at Programa STIC, Fundacion; and Dr.
+	  Manuel Sadosky, Buenos Aires, Argentina ........ Merged revisions
+	  410380 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-10 13:18 +0000 [r410311]  Kinsey Moore <kmoore at digium.com>
+
+	* /, channels/chan_sip.c: AST-2014-002: chan_sip: Exit early on bad
+	  session timers request This change allows chan_sip to avoid
+	  creation of the channel and consumption of associated file
+	  descriptors altogether if the inbound request is going to be
+	  rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey
+	  Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey
+	  Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by
+	  Corey Farrell (license 5909) ........ Merged revisions 410308
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-07 22:52 +0000 [r410225]  Corey Farrell <git at cfware.com>
+
+	* /, channels/chan_sip.c: chan_sip: Fix deadlock of monlock between
+	  unload_module and do_monitor Release monlock before calling
+	  pthread_join. This ensures do_monitor cannot freeze by locking
+	  monlock during module unload. (closes issue ASTERISK-21406)
+	  Reported by: Corey Farrell Review:
+	  https://reviewboard.asterisk.org/r/3284/ ........ Merged
+	  revisions 410224 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-07 04:38 +0000 [r410106]  Matthew Jordan <mjordan at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Allow static realtime members
+	  to be qualified during module load. When a static realtime peer
+	  with qualify=yes is loaded, Asterisk will fail to send an OPTIONS
+	  request due to the lastms being equal to 0. This results in the
+	  peer being unable to receive calls from Asterisk because the
+	  status is permanently UNKNOWN. This patch allows an OPTIONS
+	  request to be sent during module load by ignoring the lastms
+	  value on startup only. Review:
+	  https://reviewboard.asterisk.org/r/3294/ (closes issue
+	  ASTERISK-17523) Reported by: Maciej Krajewski Tested by:
+	  wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor
+	  Peirce (license 6112) ........ Merged revisions 410105 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-06 23:15 +0000 [r410044]  Russell Bryant <russell at russellbryant.com>
+
+	* /, res/res_musiconhold.c: moh: fix a refcount error with realtime
+	  MOH I observed a crash in res_musiconhold on an Asterisk 11
+	  system using realtime MOH. Investigation of the backtrace showed
+	  a corrupt mohclass, implying that it got destroyed before the
+	  code expected it to. I went looking for reference counting errors
+	  that could have caused this crash and this patch this result. It
+	  contains 2 changes. 1) Remove a usless block of code that was
+	  impossible to reach. There was even a comment indicating that it
+	  was impossible to reach. The conditional includes
+	  "!ast_test_flag(global_flags, MOH_CACHERTCLASSES)" and it's
+	  inside of an if block with the opposite check
+	  "ast_test_flag(global_flags, MOH_CACHERTCLASSES)". There's no
+	  good reason to keep it around. 2) A similar block to #1 contained
+	  a reference counting error. It stores state->class in the local
+	  variable mohclass without increasing its reference count. The
+	  reference count on mohclass is decremented at the end of the
+	  function. This block of code probably very rarely runs, which
+	  would help explain why this system was working fine for many
+	  months before experiencing a crash. Review:
+	  https://reviewboard.asterisk.org/r/3282/ ........ Merged
+	  revisions 410043 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-06 01:58 +0000 [r409990]  Matthew Jordan <mjordan at digium.com>
+
+	* res/res_fax_spandsp.c: res_fax_spandsp: Fix crash when passing
+	  ulaw/alaw data to spandsp When acting as a T.38 fax gateway,
+	  res_fax_spandsp would at times cause a crash in libspandsp. This
+	  would occur when, during fax tone detection, a ulaw/alaw frame
+	  would be passed to modem_connect_tones_rx. That particular
+	  routine expects the data to be in slin format. This patch looks
+	  at the frame type and, if the data is ulaw/alaw, converts the
+	  format to slin before passing it to modem_connect_tones_rx.
+	  Review: https://reviewboard.asterisk.org/r/3296 (closes issue
+	  ASTERISK-20149) Reported by: Alexandr Gordeev Tested by: Michal
+	  Rybarik patches: spandsp_g711decode.diff uploaded by Michal
+	  Rybarik (license 6578)
+
+2014-03-05 20:37 +0000 [r409917]  Kinsey Moore <kmoore at digium.com>
+
+	* main/config.c, /: config: Fix inverted test The test of the
+	  result of the stat() call was inverted such that its output was
+	  only used if the call failed. This inverts the test so that the
+	  output of stat() is used correctly. This was causing full reloads
+	  on unchanged files. (closes issue ASTERISK-23383) Reported by:
+	  David Woolley ........ Merged revisions 409916 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 18:45 +0000 [r409886]  Mark Michelson <mmichelson at digium.com>
+
+	* funcs/func_presencestate.c: Fix documentation for PRESENCE_STATE
+	  to properly illustrate how to create a presence hint. There was a
+	  missing comma. This was discovered by Dan Kaplan.
+
+2014-03-05 16:55 +0000 [r409834]  David M. Lee <dlee at digium.com>
+
+	* main/config.c, /, configure, include/asterisk/autoconfig.h.in,
+	  configure.ac: Corrected cross-platform stat nanosecond code When
+	  nanosecond time resolution was added for identifying config file
+	  changes, it didn't cover all of the myriad of ways that one might
+	  obtain nanosecond time resolution off of struct stat. Rather than
+	  complicate the #if even further figuring out one system from the
+	  next, this patch directly tests for the three struct members I
+	  know about today, and #ifdef's accordingly. Review:
+	  https://reviewboard.asterisk.org/r/3273/ ........ Merged
+	  revisions 409833 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 12:04 +0000 [r409778]  Sean Bright <sean at malleable.com>
+
+	* /, contrib/scripts/astgenkey, contrib/scripts/astgenkey.8: Fix
+	  references to 'keys' CLI commands in astgenkey ........ Merged
+	  revisions 409777 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 06:28 +0000 [r409745-409761]  Igor Goncharovskiy <igor.goncharovsky at gmail.com>
+
+	* channels/chan_unistim.c, /: Correct RTP handling in chan_unistim
+	  and fix transfer process broken in previous fix: - Fixed too
+	  early RTP setup with phone, that cause no ringback tone on caller
+	  side - Handle call transfer cancel only in STATE_CALL case
+	  (related to ASTERISK-23073) (Reported by: Németh Tamás, niurkin
+	  sil)
+
+	* channels/chan_unistim.c: Add update_peer function to
+	  unistim_rtp_glue, improve other unistim_rtp_glue functions
+	  conforming to other channel drivers. Do not forget auto-detected
+	  and user-selected phone settings on 'unistim reload' ........
+	  Merged revisions 409705 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-05 04:55 +0000 [r409681-409703]  Moises Silva <moises.silva at gmail.com>
+
+	* res/res_http_websocket.c: Fix res/res_http_websocket.c build
+	  failure in 32bit due to incorrect print format for uint64_t
+
+	* res/res_http_websocket.c: Fix WebRTC over WSS not working Several
+	  fixes for the WebSockets implementation in
+	  res/res_http_websocket.c * Flush the websocket session FILE* as
+	  fwrite() may not actually guarantee sending the data to the
+	  network. If we do not flush, it seems that buffering on the SSL
+	  socket for outbound messages causes issues * Refactored
+	  ast_websocket_read to take into account that SSL file descriptors
+	  may be ready to read via fread() but poll() will not actually say
+	  so because the data was already read from the network buffers and
+	  is now in the libc buffers (closes issue ASTERISK-23099) (closes
+	  issue ASTERISK-21930) Review:
+	  https://reviewboard.asterisk.org/r/3248/
+
+2014-03-04 19:33 +0000 [r409625]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* funcs/func_audiohookinherit.c, /: func_audiohookinheritance:
+	  Check If A Channel Was Specified This patch prevents a crash when
+	  using the function audiohookinheritance without setting the
+	  channel. (closes issue ASTERISK-23104) Reported by: Joel Vandal
+	  Tested by: Joel Vandal Patches:
+	  asterisk-23104_audiohook_inherit_no_channel-11.diff uploaded by
+	  Michael L. Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/3272/ ........ Merged
+	  revisions 409623 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 16:51 +0000 [r409567]  Kinsey Moore <kmoore at digium.com>
+
+	* main/astobj2.c, /: AO2: Add an assert for bad objects This adds
+	  an assert that will only be active if Asterisk is compiled with
+	  DO_CRASH and allows the testsuite to fail tests that would
+	  otherwise require log file parsing. ........ Merged revisions
+	  409566 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 16:40 +0000 [r409565]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Fix one way audio
+	  problems with hold/unhold when using ICE ICE sessions will now be
+	  restarted if sessions are changed to use new sets of remote
+	  candidates. (closes issue ASTERISK-22911) Reported by: Vytis
+	  Valentinavičius Review: https://reviewboard.asterisk.org/r/3275/
+
+2014-03-04 15:35 +0000 [r409524]  Kinsey Moore <kmoore at digium.com>
+
+	* main/rtp_engine.c, /: rtp_engine: Clean up after a failed remote
+	  bridge Upon failure of an INVITE transaction meant to initiate a
+	  remote native bridge, rtp_engine.c would not clean up
+	  non-reference-counted bridge instance pointers leaving a dangling
+	  pointer which was being used to perform a local native bridge
+	  after the other channel had hung up. This lead to dereferencing
+	  into freed memory and plenty of AO2 errors. This change allows
+	  the remote native bridge loop to clean up properly when the
+	  bridge fails. (closes issue ASTERISK-23310) Reported by: Jeremy
+	  Laine ........ Merged revisions 409521 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-04 14:52 +0000 [r409473]  Sean Bright <sean at malleable.com>
+
+	* /, channels/chan_sip.c: Minor whitespace change to 'sip show
+	  peers' output. (closes issue ASTERISK-23406) Reported by: ibercom
+	  Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom
+	  ........ Merged revisions 409472 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-03 02:07 +0000 [r409362]  Matthew Jordan <mjordan at digium.com>
+
+	* /, main/asterisk.c: doxygen: Tweak the link back to ye olde
+	  Digium website ........ Merged revisions 409361 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-03-02 12:26 +0000 [r409344]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* /, Makefile.rules: Makefile: replace -O6 with -O3 -O6 is not a
+	  legal option of gcc. Unofficially gcc considers it to be
+	  equivalent of -O3. clang chalks on it, though. This commit sets
+	  the default optimization flag to be -O3, like gcc actually
+	  considered it. Review: https://reviewboard.asterisk.org/r/3280/
+	  ........ Merged revisions 409308 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-28 21:30 +0000 [r409255]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Add precautionary p->owner
+	  checks. * Add precautionary p->owner checks in sip_hangup(),
+	  get_refer_info(), get_also_info(), and
+	  interpret_t38_parameters(). * Simplify some tangled logic in
+	  get_refer_info(), get_also_info(), and add_rpid(). * Removed some
+	  dead code in handle_request_invite(). (closes issue
+	  ASTERISK-23323) Reported by: Walter Doekes Patches:
+	  issueA23323-more_p_owner_checks-1.8.x.patch (license #5674)
+	  uploaded by wdoekes (modified)
+	  issueA23323-more_p_owner_checks-11.x.patch (license #5674)
+	  uploaded by wdoekes (modified)
+	  issueA23323-more_p_owner_checks-12.x.patch (license #5674)
+	  uploaded by wdoekes (modified)
+	  issueA23323-more_p_owner_checks-trunk.patch (license #5674)
+	  uploaded by wdoekes (modified) ........ Merged revisions 409207
+	  from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-28 21:13 +0000 [r409208]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_queue.c: app_queue: Fix documentation generation The
+	  documentation for QueueMemberPaused was causing documentation
+	  generation to fail because the documentation for that AMI event
+	  was in the wrong location. This moves that documentation the
+	  correct location and adds a missing parameter. (closes issue
+	  SWDAT-261)
+
+2014-02-28 18:00 +0000 [r409157]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_sip.c: chan_sip: Fix crash in
+	  ast_channel_hangupcause_set(). * Fix crash in
+	  ast_channel_hangupcause_set() because p->owner not checked before
+	  calling. Regression introduced by the fix for ASTERISK-22621.
+	  (closes issue ASTERISK-23135) Reported by: OK (issue
+	  ASTERISK-23323) Reported by: Walter Doekes ........ Merged
+	  revisions 409156 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 19:38 +0000 [r409129-409130]  Jonathan Rose <jrose at digium.com>
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: correct build error
+	  from r409129 Accidentally placed a declaration below functional
+	  code (issue ASTERISK-23213) Reported by: Andrea Suisani Review:
+	  https://reviewboard.asterisk.org/r/3256/
+
+	* res/res_rtp_asterisk.c: res_rtp_asterisk: Fix checklist creating
+	  problems in ICE sessions Prior to this patch, local candidate
+	  lists including SRFLX would fail to start properly when building
+	  ICE candidate check lists. This patch fixes that problem by
+	  making sure that each SRFLX candidate is associated with the
+	  proper base address so that the check list can create matches
+	  properly. This patch was written by jcolp. The issue will be left
+	  open to await testing by the issue participants. (issue
+	  ASTERISK-23213) Reported by: Andrea Suisani Review:
+	  https://reviewboard.asterisk.org/r/3256/
+
+2014-02-27 16:24 +0000 [r409083]  David M. Lee <dlee at digium.com>
+
+	* /, utils/astman.c: Fix memory stomping bug in astman. This memset
+	  complained in dev mod on my Ubuntu box. The memset is both
+	  unnecessary and dangerous. At this point, m hasn't been
+	  initialized yet, so the memset will write off to whatever address
+	  happens to be on the stack at the time. ........ Merged revisions
+	  409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 16:03 +0000 [r409053]  Corey Farrell <git at cfware.com>
+
+	* /, res/res_fax.c, configs/res_fax.conf.sample: res_fax: Warn that
+	  minrate=2400 is not valid for V.27 instead of failing load.
+	  Change minrate from 2400 to 4800 on config reload in response to
+	  changes from ASTERISK-22790 only. Any config with minrate of 2400
+	  that would fail before r405693 will still fail. Comment out many
+	  settings in res_fax.conf.sample. The defaults are set in
+	  res_fax.c, so setting the same value in sample config does
+	  nothing but make the sample config more fragile. (closes issue
+	  ASTERISK-23231) Reported by: David Brillert Review:
+	  https://reviewboard.asterisk.org/r/3261/ ........ Merged
+	  revisions 409052 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-27 12:47 +0000 [r409002]  Matthew Jordan <mjordan at digium.com>
+
+	* main/rtp_engine.c, /, include/asterisk/rtp_engine.h: rtp_engine:
+	  fix crash during remote native bridging when calling get_codecs
+	  When two RTP channels are in a remote bridge, the remote bridging
+	  loop in rtp_engine will periodically check to see if the two
+	  channels can still be bridged. One of the many things it checks
+	  is whether or not the codecs have changed on the channel. If the
+	  codec has changed, it will break out of the loop to re-determine
+	  which type of bridge is appropriate. In order to perform this
+	  check, the ast_rtp_glue virtual table's get_codec callback is
+	  called for each channel. The callback implementations assume that
+	  the channel tech private is valid when called; as such, there has
+	  always been some code in place to check whether or not the
+	  channel pvt is NULL before calling. However, this check is
+	  insufficient. The channels are unlocked during the remote
+	  bridging loop. It is possible for a channel to get masqueraded
+	  between the check for the pvt being NULL and the actual call to
+	  get_codec. When this occurs, the callback is called with a ZOMBIE
+	  channel, which now has a NULL pvt. Crash. While this has always
+	  been possible in Asterisk 1.8, it is much more likely to occur in
+	  Asterisk 11 and later versions due to the timing changes that
+	  occur when getting the codec from a channel. Note that this is
+	  much more likely to be reproduced on slow, boggy hardware running
+	  Asterisk 11 - but fairly rarely otherwise. Also Note: This crash
+	  was also caught by the various SIP blind transfer tests, in
+	  addition to the bug report Alec filed. Review:
+	  https://reviewboard.asterisk.org/r/3247/ (closes issue
+	  ASTERISK-21737) Reported by: Alec Davis Tested by: Alec Davis
+	  ........ Merged revisions 409001 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-25 17:43 +0000 [r408877]  Rusty Newton <rnewton at digium.com>
+
+	* /, configs/voicemail.conf.sample: configs/voicemail.conf.sample -
+	  Make mailcmd sample text more explicit Made the wording a bit
+	  more explicit. Didn't really change the meaning. ........ Merged
+	  revisions 408876 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-22 17:42 +0000 [r408838]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/chan_ooh323.c: ignore AST_CONTROL_PVT_CAUSE_CODE without
+	  any messages (closes issue ASTERISK-23336) Reported by: Alexander
+	  Semych
+
+2014-02-22 02:28 +0000 [r408786]  Corey Farrell <git at cfware.com>
+
+	* res/ael/pval.c, main/pbx.c, /, utils/extconf.c, utils/conf2ael.c:
+	  Remove extra defines of AST_PBX_MAX_STACK. * Ensure
+	  AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix
+	  incorrect function parameters in utils/extconf.c. (closes issue
+	  ASTERISK-23141) Reported by: Maxim Review:
+	  https://reviewboard.asterisk.org/r/3241/ ........ Merged
+	  revisions 408785 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 20:21 +0000 [r408643-408748]  Kevin Harwell <kharwell at digium.com>
+
+	* /, apps/app_forkcdr.c: app_forkcdr: ForkCDR v option does not
+	  keep CDR variables for subsequent records When the 'v' option is
+	  specified to ForkCDR application, AST_CDR_FLAG_KEEP_VARS flag is
+	  set only for the first CDR in the chain. So ForkCDR works fine
+	  with this option only once. After the second and further calls to
+	  ForkCDR, CDR variables get cleared on all CDRs besides the first
+	  one and moved to the newly forked CDR. It always sets the
+	  KEEP_VARS flag on the first CDR in the chain, instead of the most
+	  recent CDR which is used as a base to fork a new CDR. This patch
+	  sets KEEP_VARS flag on the most recent CDR on the stack (the CDR
+	  used for forking). (closes issue ASTERISK-23260) Reported by:
+	  zvision Patches: app_forkcdr.diff uploaded by zvision (license
+	  5755) ........ Merged revisions 408747 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* contrib/scripts/install_prereq: install_prereq: Missing
+	  uuid[-dev] for debian distros Added uuid and uuid-dev to install
+	  prereq script. (closes issue ASTERISK-23255) Reported by: Rusty
+	  Newton
+
+	* contrib/scripts/install_prereq, main/rtp_engine.c: rtp_engine:
+	  Dynamic payload change in rtp mapping not supported Asterisk
+	  didn't support the dynamic payload change in rtp mapping in the
+	  200 OK response. Scenario: Asterisk sends the INVITE proposing
+	  alaw and telephone-event, it proposes rtpmap:101 for
+	  telephone-event. Peer responds with 2xx, it answers with alaw and
+	  telephone-event also, but it proposes a different rtpmap number
+	  (rtpmap:103) for telephone-event. Expected Behaviour: Asterisk
+	  should honour the rtpmapping in the response and send DTMF
+	  packets using 103 as payload type for DTMF. Actual Behaviour:
+	  Asterisk sends DTMF packets using payload type 101. With this
+	  patch asterisk now supports changes that can occur in the rtp
+	  mapping in the response. (closes issue ASTERISK-23279) Reported
+	  by: NITESH BANSAL Review:
+	  https://reviewboard.asterisk.org/r/3225/ Patches:
+	  dynamic_payload_change.patch uploaded by nbansal (license 6418)
+
+	* main/rtp_engine.c, /: rtp_engine: Output mixup in
+	  ${CHANNEL(rtpqos,audio,all)} Fixed the output of
+	  CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter.
+	  (closes issue ASTERISK-23261) Reported by: rsw686 Patches:
+	  rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged
+	  revisions 408646 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* /, channels/chan_sip.c, main/channel.c: channel.c: MOH is not
+	  working for transferee after attended transfer Updated the code
+	  to check to see if MOH is playing on the transferor and if so
+	  then start it on the channel that replaces it during a
+	  masquerade. Example scenario of the problem: Alice calls Bob and
+	  then Bob begins the attended transfer process into a queue. Upon
+	  going on hold Alice hears music and so does Bob once he is in the
+	  queue. Bob then transfers Alice into the queue and then music for
+	  Alice stops even though she should be hearing it since has now
+	  replaced Bob in the queue. The problem that was occurring is that
+	  once the channel was masqueraded the app (queues, confbridge,
+	  etc...) had no way of knowing that the channel had just been
+	  swapped out thus it did not start music for the present channel.
+	  Credit to Olle Johansson for pointing me in the right direction
+	  on this issue. (closes issue ASTERISK-19499) Reported by: Timo
+	  Teräs Review: https://reviewboard.asterisk.org/r/3226/ ........
+	  Merged revisions 408642 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 10:40 +0000 [r408590]  Alexandr Anikin <may at telecom-service.ru>
+
+	* /, addons/ooh323c/src/ooCalls.h: Fix type of roundTripDelay
+	  variables ........ Merged revisions 408589 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-21 00:47 +0000 [r408537]  Michael L. Young <elgueromexicano at gmail.com>
+
+	* /, apps/app_chanspy.c: app_chanspy: Documentation Update To
+	  Clarify "x" Option When using the "x" option (specify a DTMF
+	  digit to exit the application), it is not obvious in the
+	  documentation that this only works when spying on a channel. If a
+	  channel being used to spy on other channels is waiting to connect
+	  to a channel or is no longer attached to a channel, the DTMF is
+	  ignored. As noted on the issue tracker, since there are
+	  workarounds available and this is a rarely used option we are
+	  opting for a documentation change here. (closes issue
+	  ASTERISK-22661) Reported by: Chris Hillman Patches:
+	  asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L.
+	  Young (license 5026) Review:
+	  https://reviewboard.asterisk.org/r/2990/ ........ Merged
+	  revisions 408536 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-20 02:41 +0000 [r408448]  Rusty Newton <rnewton at digium.com>
+
+	* apps/app_queue.c, /: apps/app_queue - Fix incorrect Macro
+	  parameter documentation Macro is executed on the called channel,
+	  not the calling channel. (closes issue ASTERISK-23069) Reported
+	  By: Bryan Anderson ........ Merged revisions 408447 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-19 19:05 +0000 [r408388]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/config.c, /: config: Add file size and nanosecond resolution
+	  fields to the cached modified config file information. Repeatedly
+	  modifying config files and reloading too fast sometimes fails to
+	  reload the configuration because the cached modification
+	  timestamp has one second resolution. * Added file size and
+	  nanosecond resolution fields to the cached config file
+	  modification timestamp information. Now if the file size changes
+	  or the file system supports nanosecond resolution the modified
+	  file has a better chance of being detected for reload. * Added a
+	  missing unlock in an off-nominal code path. (closes issue
+	  AST-1303) Review: https://reviewboard.asterisk.org/r/3235/
+	  ........ Merged revisions 408387 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+2014-02-19 11:45 +0000 [r408312-408330]  Alexandr Anikin <may at telecom-service.ru>
+
+	* addons/ooh323c/src/ooCapability.c, /,
+	  addons/ooh323c/src/ooh245.c: process receiveAndTransmit user
+	  input remote caps instead of receive only send receiveAndTransmit
+	  user input our caps instead of receive only ........ Merged
+	  revisions 408328 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.8
+
+	* addons/ooh323c/src/ooh323.c: Allow different socket and
+	  signalling ip on h.323 connection if gk mode is active Reported
+	  by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by:
+	  Gabriele Odone
+
+2014-02-16 03:15 +0000 [r408193-408201]  Matthew Jordan <mjordan at digium.com>
+
+	* main/pbx.c, /: pbx: Handle a completely empty dialplan during a
+	  context merge It is highly unlikely, but - at least in Asterisk
+	  12 - theoretically possible to load Asterisk with no dialplan
+	  whatsoever. If that occurs, and some other module (that is not a
+	  pbx module) attempts to merge its contexts into the dialplan, the
+	  existing merge routine will crash. This is because it is not
+	  insane, and rightly believes that you provided some sort of
+	  dialplan, somewhere. This patch will gracefully merge the
+	  contexts in such a case. Note that this is highly unlikely to

[... 28843 lines stripped ...]



More information about the asterisk-commits mailing list