[asterisk-commits] lmadsen: tag 1.4.35-rc1 r278698 - /tags/1.4.35-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Jul 22 14:27:59 CDT 2010


Author: lmadsen
Date: Thu Jul 22 14:27:54 2010
New Revision: 278698

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=278698
Log:
Importing files for 1.4.35-rc1 release.

Added:
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    tags/1.4.35-rc1/.version   (with props)
    tags/1.4.35-rc1/ChangeLog   (with props)

Added: tags/1.4.35-rc1/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.4.35-rc1/.lastclean?view=auto&rev=278698
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==============================================================================
--- tags/1.4.35-rc1/ChangeLog (added)
+++ tags/1.4.35-rc1/ChangeLog Thu Jul 22 14:27:54 2010
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+2010-07-22  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.35-rc1 Released.
+
+2010-07-22 14:55 +0000 [r278618]  Mark Michelson <mmichelson at digium.com>
+
+	* main/channel.c: Allow PLC to function properly when channels use
+	  SLIN for audio. If a channel involved in a bridge was using SLIN
+	  audio, then translation paths were not guaranteed to be set up
+	  properly since in all likelihood the number of translation steps
+	  was only 1. This patch enforces the transcode_via_slin behavior
+	  if transcode_via_slin or generic_plc is enabled and one of the
+	  formats to make compatible is SLIN. AST-352
+
+2010-07-20 22:23 +0000 [r278023-278261]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Delete IMAP messages in reverse order, to
+	  ensure reordering after each expunge does not cause deletion of
+	  the wrong message. (closes issue #16350) Reported by: noahisaac
+	  Patches: 20100623__issue16350.diff.txt uploaded by tilghman
+	  (license 14)
+
+	* main/autoservice.c, res/res_features.c,
+	  include/asterisk/channel.h: Do not queue up DTMF frames while a
+	  call is on hold. (Fixes ABE-2110)
+
+	* main/manager.c: Off-by-one error (closes issue #16506) Reported
+	  by: nik600 Patches: 20100629__issue16506.diff.txt uploaded by
+	  tilghman (license 14)
+
+2010-07-19 20:56 +0000 [r277944]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_sip.c: Regression with T.38 negotiation Prior to
+	  1.4.26.3 T.38 negotiation worked properly, in the case of the
+	  reporter. (issue #16852) Reported by: cfc (closes issue #16705)
+	  Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded
+	  by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa,
+	  samdell3 Review: https://reviewboard.asterisk.org/r/754/
+
+2010-07-19 20:16 +0000 [r277906]  Jean Galarneau <jgalarneau at digium.com>
+
+	* res/res_features.c: Avoid trying to pickup a parked extension
+	  before the park operation is completed. A crash could occur if
+	  the extension is picked up while the parking extension is being
+	  announced. Testing pu->notquiteyet while searching for a parked
+	  extension resolves this crash. (ABE-2418)
+
+2010-07-17 16:59 +0000 [r277738]  Tilghman Lesher <tlesher at digium.com>
+
+	* autoconf/ast_func_fork.m4, configure: Remove uclibc cross-compile
+	  triplet, as uclibc has a working fork()... it's only uclinux that
+	  does not. (closes issue #17616) Reported by: pprindeville
+
+2010-07-16 22:43 +0000 [r277625]  Tim Ringenbach <tim.ringenbach at gmail.com>
+
+	* res/res_features.c: Save and restore AST_FLAG_BRIDGE_HANGUP_DONT
+	  on attended transfer. ast_bridge_call() clears
+	  AST_FLAG_BRIDGE_HANGUP_DONT. But during an attended transfer,
+	  ast_bridge_call() is called for a second bridge on the same
+	  channel, and it clears that flag, which still needs to get set
+	  for when the original ast_bridge_call() gets control back and
+	  checks it. Review: https://reviewboard.asterisk.org/r/741
+
+2010-07-16 21:54 +0000 [r277568]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_config_pgsql.c, res/res_config_odbc.c: Since we split
+	  values at the semicolon, we should store values with a semicolon
+	  as an encoded value. (closes issue #17369) Reported by: gkservice
+	  Patches: 20100625__issue17369.diff.txt uploaded by tilghman
+	  (license 14) Tested by: tilghman
+
+2010-07-16 21:18 +0000 [r277497]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Default to no udptl error correction so that
+	  error correction will be disabled in the event that the remote
+	  end indicates that they do not support the error correction mode
+	  we requested. FAX-128
+
+2010-07-16 20:18 +0000 [r277419]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: priexclusive in chan_dahdi.conf ignored
+	  when reloading dahdi module During a reload, the priexclusive and
+	  outsignalling parameters are not read in from the config file as
+	  intended. Unfortunately, they get set to defaults as a result.
+	  This patch makes sure that they do not get set to defaults during
+	  a reload. (closes issue #17441) Reported by: mtryfoss Patches:
+	  issue17441_v1.4.patch uploaded by rmudgett (license 664)
+	  issue17441_v1.6.2.patch uploaded by rmudgett (license 664)
+	  issue17441_trunk.patch uploaded by rmudgett (license 664) Tested
+	  by: rmudgett
+
+2010-07-16 18:30 +0000 [r277327]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/pbx.c: Interpret device state AST_DEVICE_UNKNOWN as
+	  extension state AST_EXTENSION_NOT_INUSE. (closes issue #16035)
+	  Reported by: francesco_r Patches: pbx.c.patch uploaded by
+	  viniciusfontes (license 978) Tested by: francesco_r, agx, lawbar
+
+2010-07-16 18:04 +0000 [r277261]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/manager.c: If variable gotten is not set, will segfault on
+	  Solaris. (closes issue #17636) Reported by: bklang
+
+2010-07-16 17:29 +0000 [r277247]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c: For pass through DTMF tones, measure the actual
+	  duration between the begin and end packets on the wire. If it is
+	  detected to be less than AST_MIN_DTMF_DURATION, trigger dtmf
+	  emulation. AST-362
+
+2010-07-16 17:10 +0000 [r277182]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* apps/app_amd.c: Total analysis time error with SIP and silence
+	  suppression When using app_amd with SIP providers that have
+	  silence suppression on, the iTotalTime count increases
+	  exponentially. (closes issue #17656) Reported by: juls
+
+2010-07-15 13:48 +0000 [r276652]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: In a perfect world, the frame source would never
+	  be NULL. In the meantime, don't crash when it is.
+
+2010-07-14 11:49 +0000 [r276267]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/voicemail.conf.sample: Update documentation for
+	  voicemail.conf externpass option.
+
+2010-07-13 19:14 +0000 [r275994-276126]  Russell Bryant <russell at digium.com>
+
+	* res/res_features.c: Only reset a CDR that exists.
+
+	* res/res_features.c: Use chan->cdr instead of chan_cdr (just like
+	  peer->cdr instead of peer_cdr in the last commit).
+
+	* res/res_features.c: Access peer->cdr directly instead of through
+	  a saved off reference. At this point in the code, it is possible
+	  that peer_cdr may be invalid. Specifically, in the blind transfer
+	  code, CDRs are swapped between channels. So, peer_cdr is no
+	  longer == peer->cdr. The scenario that exposed a crash in this
+	  code was a blind transfer that hit the system call limit, causing
+	  the transferee channel to get destroyed after the transfer
+	  attempt failed. Even if it succeeds and this code doesn't crash,
+	  this code was still trying to reset a CDR on a channel that was
+	  now owned by a different thread, which is a BadThing(tm).
+	  (ABE-2417)
+
+2010-07-13 14:47 +0000 [r275909]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/realtime/mysql/sipfriends.sql,
+	  contrib/realtime/mysql/voicemail.sql,
+	  contrib/scripts/realtime_pgsql.sql (removed),
+	  contrib/scripts/vmdb.sql (removed),
+	  contrib/scripts/iax-friends.sql (removed),
+	  contrib/realtime/mysql/iaxfriends.sql,
+	  contrib/realtime/mysql/meetme.sql, contrib/scripts/meetme.sql
+	  (removed), contrib/realtime (added), contrib/realtime/postgresql,
+	  contrib/realtime/postgresql/realtime.sql, contrib/realtime/mysql,
+	  contrib/realtime/oracle, contrib/realtime/sqlserver,
+	  contrib/scripts/sip-friends.sql (removed): Move SQL scripts into
+	  their own database-specific directories.
+
+2010-07-12 20:34 +0000 [r275665-275773]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_meetme.c: Make user removals and traversals thread safe
+	  in meetme. Race conditions present in meetme involving the user
+	  list where a lack of locking has the potential for a user to be
+	  removed during a traversal or as in the case of the reporter
+	  after checking if the list is empty could cause a crash. Fixing
+	  this was done by convering the userlist to an ao2 container.
+	  (closes issue #17390) Reported by: Vince Review:
+	  https://reviewboard.asterisk.org/r/746/
+
+	* main/channel.c: Change ast_write to not stop generator when
+	  called from ast_prod. For SIP channels configured with the
+	  progressinband option on, the ringback was being immediately
+	  stopped. This problem was due to ast_prod being moved for a
+	  deadlock fix in 259858. Prodding the channel after setting up the
+	  generator triggered the check in ast_write to stop the generator.
+	  The fix here should write the frame the same as was done before
+	  the call to ast_prod was moved. (closes issue #17372) Reported
+	  by: tech_admin
+
+2010-07-09 19:28 +0000 [r275241-275290]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* main/cli.c: fix tab-completion for unload command. (closes issue
+	  #17536) Reported by: junky Patches: unload_vs_mod_unload.diff
+	  uploaded by junky (license 177) Tested by: pabelanger
+
+	* channels/chan_sip.c: Fix logging message for stale nonce. (closes
+	  issue #17582) Reported by: kenner Patches: chan_sip.c.diff
+	  uploaded by kenner (license 1040) Tested by: lmadsen
+
+2010-07-09 18:23 +0000 [r275027-275182]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/loader.c: give a better error message when attempting to
+	  unload a module that is not loaded
+
+	* main/loader.c: don't unload modules that returned
+	  AST_MODULE_LOAD_DECLINE when they were loaded
+
+	* apps/app_dial.c: Clear the AST_CDR_FLAG_DIALED flag for channels
+	  going into the pbx via the G option in app_dial (closes issue
+	  #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff
+	  uploaded by mnicholson (license 96) Tested by: jamicque,
+	  mnicholson
+
+2010-07-09 15:33 +0000 [r275021]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/test.h, main/test.c: Document that a leading and
+	  trailing slash is expected for test categories. Also, emit a
+	  warning if a test is registered without one of these.
+
+2010-07-07 18:12 +0000 [r274579]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Close the DAHDI FD on error when
+	  processing chan_dahdi toneduration config parameter.
+
+2010-07-07 06:13 +0000 [r274417]  Tilghman Lesher <tlesher at digium.com>
+
+	* configs/say.conf.sample: Correct how 100, 200, 300, etc. is said.
+	  Also add the crazy British numbers. (closes issue #16102)
+	  Reported by: Delvar Patches: say.conf.fix.patch uploaded by
+	  Delvar (license 908) (plus a few additional fixes and
+	  simplifications by me)
+
+2010-07-06 22:46 +0000 [r274283-274359]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/Makefile: Ensure file.o is built correctly. (related to
+	  issue #15250)
+
+	* configs/sip.conf.sample: Correct sip.conf.sample comments for
+	  prematuremedia option. (closes issue #17513) Reported by: festr
+	  Patches: patch uploaded by festr (license 443)
+
+2010-07-06 22:08 +0000 [r274280]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c, configs/sip.conf.sample: Add option to not
+	  do a call forward on 482 Loop Detected Asterisk has always set up
+	  a forwarded call when receiving a 482 Loop Detected. This
+	  prevents handling the call failure by just continuing on in the
+	  dialplan. Since this would be a change in behavior, the new
+	  option to disable this behavior is forwardloopdetected which
+	  defaults to 'yes'. Review:
+	  https://reviewboard.asterisk.org/r/764/
+
+2010-07-06 14:29 +0000 [r274157]  Mark Michelson <mmichelson at digium.com>
+
+	* main/rtp.c: Fix problem with RFC 2833 DTMF not being accepted. A
+	  recent check was added to ensure that we did not erroneously
+	  detect duplicate DTMF when we received packets out of order. The
+	  problem was that the check did not account for the fact that the
+	  seqno of an RTP stream will roll over back to 0 after hitting
+	  65535. Now, we have a secondary check that will ensure that the
+	  seqno rolling over will not cause us to stop accepting DTMF.
+	  (closes issue #17571) Reported by: mdeneen Patches:
+	  rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+	  Tested by: richardf, maxochoa, JJCinAZ
+
+2010-07-06 13:52 +0000 [r274093]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: Make get_member_status return QUEUE_NO_MEMBERS
+	  instead of QUEUE_NO_REACHABLE_MEMBERS to make joinempty=no work
+	  again. This regression was introduced in 273639. Also fixed
+	  whitespace.
+
+2010-07-05 19:48 +0000 [r273981]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_oss.c, channels/chan_iax2.c: Command 'stop
+	  gracefully' doesn't.
+
+2010-07-05 13:51 +0000 [r273884]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* main/config.c: Remove extra line breaks from 'core show config
+	  mappings' (closes issue #17583) Reported by: pabelanger Patches:
+	  issue17583.patch uploaded by pabelanger (license 224) Tested by:
+	  lmadsen
+
+2010-07-02 21:36 +0000 [r273717-273793]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_dahdi.c, channels/chan_local.c, configure,
+	  include/asterisk/autoconfig.h.in, channels/chan_agent.c,
+	  configure.ac, channels/chan_h323.c, include/asterisk/lock.h,
+	  include/asterisk/compiler.h: Have the DEADLOCK_AVOIDANCE macro
+	  warn when an unlock fails, to help catch potentially large
+	  software bugs. (closes issue #17407) Reported by: pdf Patches:
+	  20100527__issue17407.diff.txt uploaded by tilghman (license 14)
+	  Review: https://reviewboard.asterisk.org/r/751/
+
+	* main/autoservice.c: Autoservice loop optimization causes a busy
+	  loop, when channels are serviced while in hangup. (closes issue
+	  #17564) Reported by: ramonpeek Patches:
+	  20100630__issue17564.diff.txt uploaded by tilghman (license 14)
+	  Tested by: ramonpeek
+
+2010-07-02 15:54 +0000 [r273640]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* apps/app_voicemail.c, channels/chan_dahdi.c,
+	  channels/chan_misdn.c, channels/chan_sip.c, res/res_agi.c,
+	  res/res_jabber.c: Fix various typos, reported by Lintian
+
+2010-07-02 15:46 +0000 [r273639]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_queue.c: If all members are paused, the wrong status is
+	  indicated. (closes issue #17576) Reported by: ramonpeek Patches:
+	  diff.txt uploaded by ramonpeek (license 266) Tested by: ramonpeek
+
+2010-07-01 22:09 +0000 [r273565]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: Don't return a partially initialized datastore.
+	  If memory allocation fails in ast_strdup(), don't return a
+	  partially initialized datastore. Bad things may happen. (related
+	  to ABE-2415)
+
+2010-07-01 20:19 +0000 [r273354-273474]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_meetme.c: Allow admin user to join conference without
+	  using admin mode and no user pin. Configuring the conference in
+	  meetme.conf like the following: conf => 2345,,6666 did not prompt
+	  for pin when used without admin mode. This meant that the
+	  conference could not be joined as an admin even if the user knew
+	  the correct pin. The original bug report was submitted claiming
+	  that the blank user pin should deny entry into the conference. I
+	  think a better way to handle this would be with a feature
+	  enhancement that used the following syntax: conf => 2345,X,6666 -
+	  where X denotes no acceptable pin allowed (closes issue #15704)
+	  Reported by: modelnine
+
+	* apps/app_meetme.c: Ensure channel placed in meetme in ringing
+	  state is properly hung up. An outgoing channel placed in meetme
+	  while still ringing which was then hung up would not exit meetme
+	  and the channel was not properly destroyed. Specifically checking
+	  for this scenario by looking at the appropriate control frames
+	  resolves the issue. (closes issue #15871) Reported by: Ivan
+	  Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan
+	  (license 229)
+
+2010-06-29 23:15 +0000 [r273057-273060]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Allow the "useragent" value to be restored
+	  into memory from the realtime backend. This value is purely
+	  informational. It does not alter configuration at all. (closes
+	  issue #16029) Reported by: Guggemand Patches:
+	  realtime-useragent.patch uploaded by Guggemand (license 897)
+	  Tested by: Guggemand
+
+	* main/channel.c: _Really_ skip the channel... don't just retry for
+	  another 200 cycles. (Closes issue SWP-1652, ABE-2240)
+
+2010-06-29 21:36 +0000 [r273017]  Russell Bryant <russell at digium.com>
+
+	* /: Remove properties that were erroneously merged to 1.4 from one
+	  of my branches.
+
+2010-07-22  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.34 Released.
+
+2010-07-07  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.34-rc2 Released.
+
+	* Fix problem with RFC 2833 DTMF not being accepted.
+  
+	  A recent check was added to ensure that we did not erroneously
+	  detect duplicate DTMF when we received packets out of order.
+	  The problem was that the check did not account for the fact that
+	  the seqno of an RTP stream will roll over back to 0 after hitting
+	  65535. Now, we have a secondary check that will ensure that the
+	  seqno rolling over will not cause us to stop accepting DTMF.
+  
+	  (closes issue 0017571)
+	  Reported by: mdeneen
+	  Patches:
+	        rtp_seqno_rollover.patch uploaded by mmichelson (license 60)
+	  Tested by: richardf, maxochoa, JJCinAZ
+
+	* Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx
+	  via the G option in app_dial
+  
+	  (closes issue 0017592)
+	  Reported by: jamicque
+	  Patches:
+	        G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
+	  Tested by: jamicque, mnicholson
+
+2010-06-29  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.34-rc1 Released.
+
+2010-06-28 21:50 +0000 [r272921-272925]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c: Don't change ownership/group/permissions on run
+	  directory, if it already exists. (closes issue #17076) Reported
+	  by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by
+	  tilghman (license 14) Tested by: stuarth
+
+	* main/config.c: Also trim trailing blanks on #includes
+
+	* main/config.c: Change the way that we read include files, to
+	  accommodate for changes in GCC 4.4. (closes issue #17472)
+	  Reported by: seandarcy Patches: config2.patch uploaded by nivan
+	  (license 1066) Tested by: nivan
+
+2010-06-28 18:47 +0000 [r272878-272881]  Russell Bryant <russell at digium.com>
+
+	* tests/test_astobj2.c (added): Backport applicable parts of
+	  test_astobj2.
+
+	* main/asterisk.c, Makefile, include/asterisk/test.h (added),
+	  build_tools/cflags-devmode.xml, include/asterisk.h,
+	  tests/Makefile, tests/test_skel.c, /, main/Makefile, tests
+	  (added), include/asterisk/linkedlists.h, main/test.c (added):
+	  Backport unit test API to 1.4. Review:
+	  https://reviewboard.asterisk.org/r/750/
+
+2010-06-28 17:31 +0000 [r272804]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Decode URI in contact header of 302
+	  response. ABE-2352
+
+2010-06-28 17:11 +0000 [r272688-272763]  Russell Bryant <russell at digium.com>
+
+	* Makefile: Force SILENTMAKE where it is needed.
+
+	* Makefile: Backport method of setting SUBMAKE from trunk. By
+	  setting the PRINT_DIR variable, SUBMAKE will print out the
+	  directories it descends into, which is important for editors
+	  (like vim) that watch the build output so that they can take you
+	  to the file where an error occurred.
+
+2010-06-25 20:17 +0000 [r272562]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/voicemail_odbc_postgresql.txt: Make the structure of the
+	  table specified before match the queries and results. (closes
+	  issue #17557) Reported by: cmaj
+
+2010-06-24 21:58 +0000 [r272446]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: ss_thread calls pri_grab without lock
+	  during overlap dial Recent changes to chan_dahdi with relation to
+	  overlap dialing call pri_grab without first obtaining a lock.
+	  (closes issue #17414) Reported by: pdf Patches: bug17414.patch
+	  uploaded by jpeeler (license 325)
+
+2010-06-23 22:33 +0000 [r272367]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: Send AgentComplete manager events in the event
+	  of blind and attended transfers. (closes issue #16819) Reported
+	  by: elbriga Patches: app_queue.diff uploaded by elbriga (license
+	  482)
+
+2010-06-23 20:57 +0000 [r272255]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* apps/app_meetme.c: First caller into a dynamic conference now
+	  enter pin once. If MeetMe is configured to use dynamic conference
+	  numbers, then the first caller (which creates the conference) had
+	  to enter the PIN number twice. (closes issue #15878) Reported by:
+	  shawkris Patches: issue15878.patch uploaded by pabelanger
+	  (license 224) Tested by: pabelanger
+
+2010-06-23 18:40 +0000 [r272147]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Backport part of revision 136715 to fix
+	  callerid in voicemail text files (IMAP only). (closes issue
+	  #16945) Reported by: mneuhauser
+
+2010-06-22 17:31 +0000 [r271689-271902]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Decrease the module ref count in sip_hangup
+	  when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep
+	  the ref count correct. (closes issue #16815) Reported by: rain
+	  Patches: chan_sip-unref-fix.diff uploaded by rain (license 327)
+	  (modified) Tested by: rain
+
+	* pbx/pbx_dundi.c: Allow users to specify a port for dundi peers.
+	  (closes issue #17056) Reported by: klaus3000 Patches:
+	  dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
+	  Tested by: klaus3000
+
+	* configs/sip_notify.conf.sample, channels/chan_sip.c: Modify
+	  chan_sip's packet generation api to automatically calculate the
+	  Content-Length. This is done by storing packet content in a
+	  buffer until it is actually time to send the packet, at which
+	  time the size of the packet is calculated. This change was made
+	  to ensure that the Content-Length is always correct. (closes
+	  issue #17326) Reported by: kenner Tested by: mnicholson, kenner
+	  Review: https://reviewboard.asterisk.org/r/693/
+
+2010-06-21 20:37 +0000 [r271552]  Jeff Peeler <jpeeler at digium.com>
+
+	* pbx/pbx_ael.c: Do not use sizeof to calculate size of a heap
+	  allocated character array. Change left out from 271399. (closes
+	  issue #16053) Reported by: diLLec
+
+2010-06-18 20:52 +0000 [r271399-271444]  Jeff Peeler <jpeeler at digium.com>
+
+	* pbx/pbx_ael.c: Check for newly added memory allocation failures
+	  gracefully during AEL2 parsing.
+
+	* pbx/pbx_ael.c: Fix crash when parsing some heavily nested
+	  statements in AEL on reload. Due to the recursion used when
+	  compiling AEL in gen_prios, all the stack space was being
+	  consumed when parsing some AEL that contained nesting 13 levels
+	  deep. Changing a few large buffers to be heap allocated fixed the
+	  crash, although I did not test how many more levels can now be
+	  safely used. (closes issue #16053) Reported by: diLLec Tested by:
+	  jpeeler
+
+2010-06-18 18:54 +0000 [r271339-271340]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/lock.h: Remove an unnecessary assignment that
+	  causes a DEBUG_THREADS build failure on mac os x.
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  include/asterisk/lock.h: Fix a build problem on Mac OS X with
+	  DEBUG_THREADS enabled. This set of changes was already in trunk.
+
+2010-06-18 18:33 +0000 [r271335]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Eliminate deadlock potential in
+	  dahdi_fixup(). (This is a backport of 269307, committed to trunk
+	  by rmudgett.) Calling dahdi_indicate() when the channel private
+	  lock is already held can cause a deadlock if the PRI lock is
+	  needed because dahdi_indicate() will also get the channel private
+	  lock. The pri_grab() function assumes that the channel private
+	  lock is held once to avoid deadlock. (closes issue #17261)
+	  Reported by: aragon
+
+2010-06-22  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.33.1 Released.
+
+	* channels/chan_dahdi.c: Merge revision 270404 from the 1.4 branch.
+
+	  fixes FXS port still ringing when answered, as reported by Tzafrir
+	  on dev-list.
+
+	  (issue #17067)
+	  Reported by: tzafrir
+	  Tested by: alecdavis
+
+2010-06-17  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.33 Released.
+
+2010-06-10  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.33-rc2 Released.
+
+2010-06-10  Tilghman Lesher <tlesher at digium.com>
+
+	* Ensure signals are not blocked inside other signal handlers.
+
+	  This eliminates the annoying <beep> on the console.
+
+	  (closes issue 0017477)
+	   Reported by: jvandal
+	   Patches:
+	         20100610__issue17477.diff.txt uploaded by tilghman (license 14)
+
+2010-06-09  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* Fix Debian init script to not use -c.
+
+	  When using the init script as-is currently, it could cause issues on Debian
+	  such as high CPU usage. This fix has worked for several people so I'm
+	  implementing the change. We now handle color displays properly.
+
+	  (closes issue 0016784)
+	  Reported by: pabelanger
+	  Patches:
+	        20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
+	  Tested by: pabelanger, tilghman
+
+2010-06-01  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.33-rc1 Released.
+
+2010-06-01 15:17 +0000 [r266585]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c: Prevent CLI prompt from distorting output of
+	  lines shorter than the prompt. Uses the VT100 method of clearing
+	  the line from the cursor position to the end of the line: Esc-0K
+	  (closes issue #17160) Reported by: coolmig Patches:
+	  20100531__issue17160.diff.txt uploaded by tilghman (license 14)
+	  Tested by: coolmig
+
+2010-06-01 14:57 +0000 [r266579-266580]  Paul Belanger <paul.belanger at polybeacon.com>
+
+	* channels/chan_sip.c: Fix formatting issue with previous patch.
+
+	* channels/chan_sip.c: Missing fallback to audio fax feature when
+	  T.38 re-INVITE failed When a T.38 re-INVITE failed with an 488 or
+	  606 answer, we should fallback to audio fax by send a
+	  re-re-INVITE without T.38. The function is backported from 1.6
+	  asterisk. (closes issue #16795) Reported by: vrban (closes issue
+	  #16692) Reported by: vrban Patches:
+	  t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
+	  Tested by: lmadsen, vrban, haggard
+	  https://reviewboard.asterisk.org/r/514/
+
+2010-05-30 04:43 +0000 [r266437]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/rc.debian.asterisk: Reverting patch and reopening
+	  issue #16784, as patch breaks color display.
+
+2010-05-26 21:11 +0000 [r266142]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, main/logger.c: Use sigaction for signals which
+	  should persist past the initial trigger, not signal. If you call
+	  signal() in a Solaris signal handler, instead of just resetting
+	  the signal handler, it causes the signal to refire, because the
+	  signal is not marked as handled prior to the signal handler being
+	  called. This effectively causes Solaris to immediately exceed the
+	  threadstack in recursive signal handlers and crash. (closes issue
+	  #17000) Reported by: rmcgilvr Patches:
+	  20100526__issue17000.diff.txt uploaded by tilghman (license 14)
+	  Tested by: rmcgilvr
+
+2010-05-26 20:33 +0000 [r266140]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c: add dahdi_func_write to zap_tech structure
+	  This was supposed to be committed with r263292, the back-port of
+	  teh DAHDI buffer policy dial string option
+
+2010-05-26 18:21 +0000 [r266004]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Make AgentComplete message more consistent. At
+	  times, the "Member" field was not specified during the event.
+	  It's there now. (closes issue #15638) Reported by: elbriga
+	  Patches: patchAppQueueAgentComplete.diff uploaded by elbriga
+	  (license 482)
+
+2010-05-26 16:21 +0000 [r265910]  Tilghman Lesher <tlesher at digium.com>
+
+	* res/res_config_pgsql.c: Not finding rows in the DB does not rise
+	  to the level of a warning. (closes issue #17062) Reported by:
+	  drookie Patches: 20100525__issue17062.diff.txt uploaded by
+	  tilghman (license 14)
+
+2010-05-25 17:11 +0000 [r265613]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c: fixes build issue with zaptel (closes
+	  issue #17394) Reported by: aragon Patches: half_buffer_fix.diff
+	  uploaded by dvossel (license 671) Tested by: aragon
+
+2010-05-25 16:48 +0000 [r265610]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: Don't mark the cdr records of unanswered queue
+	  calls with "NOANSWER". This restores the behavior prior to
+	  r258670. (closes issue #17334) Reported by: jvandal Patches:
+	  queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested
+	  by: aragon, jvandal
+
+2010-05-25 13:33 +0000 [r265570]  Terry Wilson <twilson at digium.com>
+
+	* include/asterisk/options.h, main/asterisk.c, Makefile,
+	  doc/manager.txt, main/manager.c: Merged revisions 265320,265467
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24
+	  May 2010) | 14 lines Add the FullyBooted AMI event It is possible
+	  to connect to the manager interface before all Asterisk modules
+	  are loaded. To ensure that an application does not send AMI
+	  actions that might require a module that has not yet loaded, the
+	  application can listen for the FullyBooted manager event. It will
+	  be sent upon connection if all modules have been loaded, or as
+	  soon as loading is complete. The event: Event: FullyBooted
+	  Privilege: system,all Status: Fully Booted Review:
+	  https://reviewboard.asterisk.org/r/639/ ........ r265467 |
+	  twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line
+	  Merge the rest of the FullyBooted patch ........
+
+2010-05-24 19:37 +0000 [r265365]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c: fixes segfault when using generic plc
+
+2010-05-21 20:59 +0000 [r264996-265089]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/file.h, apps/app_queue.c: Don't hang up on a
+	  queue caller if the file we attempt to play does not exist. This
+	  also fixes a documentation mistake in file.h that made my
+	  original attempt to correct this problem not work correctly.
+	  (closes issue #17061) Reported by: RoadKill
+
+	* include/asterisk/channel.h: Fix grammatical error in comment.
+
+	* main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+	  Allow ast_safe_sleep to defer specific frames until after the
+	  sleep has concluded. From reviewboard Background: A Digium
+	  customer discovered a somewhat odd bug. The setup is that parties
+	  A and B are bridged, and party A places party B on hold. While
+	  party B is listening to hold music, he mashes a bunch of DTMF.
+	  Party A takes party B off hold while this is happening, but party
+	  B continues to hear hold music. I could reproduce this about 1 in
+	  5 times. The issue: When DTMF features are enabled and a user
+	  presses keys, the channel that the DTMF is streamed to is placed
+	  in an ast_safe_sleep for 100 ms, the duration of the emulated
+	  tone. If an AST_CONTROL_UNHOLD frame is read from the channel
+	  during the sleep, the frame is dropped. Thus the unhold
+	  indication is never made to the channel that was originally
+	  placed on hold. The fix: Originally, I discussed with Kevin
+	  possible ways of fixing the specific problem reported. However,
+	  we determined that the same type of problem could happen in other
+	  situations where ast_safe_sleep() is used. Using autoservice as a
+	  model, I modified ast_safe_sleep_conditional() to defer specific
+	  frame types so they can be re-queued once the sleep has finished.
+	  I made a common function for determining if a frame should be
+	  deferred so that there are not two identical switch blocks to
+	  maintain. Review: https://reviewboard.asterisk.org/r/674/
+
+2010-05-20 23:23 +0000 [r264820]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/callerid.c: ast_callerid_parse() had a path that left name
+	  uninitialized. Several callers of ast_callerid_parse() do not
+	  initialize the name parameter before calling thus there is the
+	  potential to use an uninitialized pointer.
+
+2010-05-20 15:59 +0000 [r264541]  Mark Michelson <mmichelson at digium.com>
+
+	* include/asterisk/options.h, main/loader.c, main/channel.c,
+	  include/asterisk/channel.h: 1.4 version of PLC fix. Analogous to
+	  trunk revision 264452, but without the change to chan_sip since
+	  it is not necessary in this branch.
+
+2010-05-19 20:01 +0000 [r264334]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_speech_utils.c: Set quieted flag when receiving a dtmf
+	  tone during playback in speechbackground. (closes issue #16966)
+	  Reported by: asackheim
+
+2010-05-19 17:41 +0000 [r264248]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/options.h, configure, configure.ac: Internal
+	  timing is now on by default, if you're using DAHDI 2.3 or above.
+	  The reason for ensuring DAHDI 2.3 or above is that this version
+	  ensures that a timer is always available, whereas in previous
+	  versions, it was possible for DAHDI to be loaded, but have no
+	  drivers to actually generate timing. If internal_timing was
+	  turned on in this circumstance, a complete lack of audio would
+	  result. This is the reason why internal_timing was not on by
+	  default. However, now that DAHDI ensures the availability of a
+	  timer, there is no reason for this setting to be off (and in
+	  fact, it solves a great many initial user problems). (closes
+	  issue #15932) Reported by: dimas Patches:
+	  20100519__issue15932.diff.txt uploaded by tilghman (license 14)
+	  Tested by: tilghman
+
+2010-05-19 08:23 +0000 [r264056]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* configs/indications.conf.sample: fix incorrectly typed
+	  indications for [nz] stutter and dialrecall (closes issue #17359)
+	  Reported by: alecdavis Patches: bug17359.diff.txt uploaded by
+	  alecdavis (license 585)
+
+2010-05-19 06:32 +0000 [r263949]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/dsp.c: Because progress is called multiple times, across
+	  several frames, we must persist states when detecting multitone
+	  sequences. (closes issue #16749) Reported by: dant Patches:
+	  dsp.c-bug16749-1.patch uploaded by dant (license 670) Tested by:
+	  dant
+
+2010-05-18 18:54 +0000 [r263769]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_directory.c: Modify directory name reading to be
+	  interrupted with operator or pound escape. In the case of
+	  accidentally entering the wrong first three letters for the
+	  reading, users could be very frustrated if the name listing is
+	  very long. This allows interrupting the reading by pressing 0 or
+	  #. 0 will attempt to execute a configured operator (o) extension
+	  and # will exit and proceed in the dialplan. ABE-2200
+
+2010-05-17 22:00 +0000 [r263637-263639]  Mark Michelson <mmichelson at digium.com>
+
+	* main/devicestate.c: Fix logic error when checking for a devstate
+	  provider. When using strsep, if one of the list of specified
+	  separators is not found, it is the first parameter to strsep
+	  which is now NULL, not the pointer returned by strsep. This issue
+	  isn't especially severe in that the worst it is likely to do is
+	  waste some cycles when a device with no '/' and no ':' is passed
+	  to ast_device_state.
+
+	* main/pbx.c: Remove arbitrary size limitation for hints. (closes
+	  issue #17257) Reported by: tim_ringenbach Patches:
+	  hints_crash_fix.diff uploaded by tim ringenbach (license 540)
+
+2010-05-17 14:35 +0000 [r263374-263456]  Leif Madsen <lmadsen at digium.com>
+
+	* main/http.c: Manager cookies are not compatible with RFC2109. The
+	  Version field in the cookies we're setting contain quotes around
+	  the version number which is not compatible with RFC2109 and
+	  breaks some implementations. (closes issue #17231) Reported by:
+	  ecarruda Patches: manager_rfc2109-trunk-v1.patch uploaded by
+	  ecarruda (license 559) manager_rfc2109-1.6.2-v1.patch uploaded by
+	  ecarruda (license 559) Tested by: ecarruda, russell
+
+	* sounds/Makefile: Update link to new version of core sounds. The
+	  latest version of the core sounds files 1.4.19 now includes the
+	  missing queue-minute sound file which is called by app_queue but
+	  which has been missing. (closes issue #17123) Reported by:
+	  n8ideas
+
+2010-05-17 13:01 +0000 [r263292]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_dahdi.c: backport of DAHDI buffer policy dial
+	  string option
+
+2010-05-13 23:08 +0000 [r263112]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c, main/file.c: Fix internal timing not working with
+	  Zaptel dahdi_compat.h was not being included in channel.c when
+	  used with Zaptel and wasn't in file.c at all. (closes issue
+	  #15250) Reported by: mneuhauser Patches: dahdi_compat.patch
+	  uploaded by mneuhauser (license 425) Tested by: IgorG
+
+2010-05-12 17:00 +0000 [r262662]  David Vossel <dvossel at digium.com>
+
+	* apps/app_meetme.c: fixes app_meetme dsp error We attempted to
+	  detect silence after translating a frame from signed linear. This
+	  caused a flooding of errors. To resolve this the code to detect
+	  silence was moved before the translation. (closes issue #17133)
+	  Reported by: jsdyer
+
+2010-05-11 19:55 +0000 [r262421]  Jason Parker <jparker at digium.com>
+

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