[asterisk-commits] lmadsen: tag 1.4.28-rc1 r233954 - /tags/1.4.28-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Dec 9 15:15:49 CST 2009


Author: lmadsen
Date: Wed Dec  9 15:15:45 2009
New Revision: 233954

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=233954
Log:
Importing files for 1.4.28-rc1 release.

Added:
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    tags/1.4.28-rc1/.version   (with props)
    tags/1.4.28-rc1/ChangeLog   (with props)

Added: tags/1.4.28-rc1/.lastclean
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--- tags/1.4.28-rc1/ChangeLog (added)
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+2009-12-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.28-rc1
+
+2009-12-09 19:58 +0000 [r233782-233879]  Russell Bryant <russell at digium.com>
+
+	* main/loader.c: Fix breakage of the "module load <module>" CLI
+	  command.
+
+	* main/loader.c, formats/format_ilbc.c, formats/format_vox.c,
+	  include/asterisk/module.h, formats/format_pcm.c,
+	  formats/format_h263.c, formats/format_g723.c,
+	  formats/format_h264.c, formats/format_jpeg.c,
+	  formats/format_g726.c, formats/format_gsm.c,
+	  formats/format_g729.c, formats/format_sln.c,
+	  formats/format_wav.c, formats/format_ogg_vorbis.c,
+	  formats/format_wav_gsm.c: Set a module load priority for format
+	  modules. A recent change to app_voicemail made it such that the
+	  module now assumes that all format modules are available while
+	  processing voicemail configuration. However, when autoloading
+	  modules, it was possible that app_voicemail was loaded before the
+	  format modules. Since format modules don't depend on anything,
+	  set a module load priority on them to ensure that they get loaded
+	  first when autoloading. This version of the patch is specific to
+	  Asterisk 1.4 and 1.6.0. These versions did not already support
+	  module load priority in the module API. This adds a trivial
+	  version of this which is just a module flag to include it in a
+	  pass before loading "everything". Thanks to mmichelson for the
+	  review! (closes issue #16412) Reported by: jiddings Tested by:
+	  russell Review: https://reviewboard.asterisk.org/r/445/
+
+2009-12-08 00:02 +0000 [r233618]  Atis Lezdins <atis at iq-labs.net>
+
+	* contrib/valgrind.supp: Merged revisions 233577 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r233577 |
+	  atis | 2009-12-08 01:10:13 +0200 (Tue, 08 Dec 2009) | 8 lines Fix
+	  compatibility with valgrind 3.3 and older. (noticed in issue
+	  #16388) Reported by: parisioa Patches: valgrind.supp uloaded by
+	  atis (license 242) Tested by: atis, parisioa ........
+
+2009-12-07 23:24 +0000 [r233471-233609]  David Vossel <dvossel at digium.com>
+
+	* main/utils.c: hex escape control and non 7-bit clean characters
+	  in uri_encode In ast_uri_encode, non 7-bit clean characters were
+	  being hex escaped correctly, but control characters were not.
+	  (issue #16299)
+
+	* channels/chan_sip.c: fixes missing Contact header angle brackets
+	  (closes issue #16298) Reported by: mgernoth Patches:
+	  reg_parse_issue_1.4.diff uploaded by dvossel (license 671) Tested
+	  by: dvossel
+
+2009-12-07 16:11 +0000 [r233392]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Allow SDP packets with only video session
+	  information. (closes issue #16387) Reported by: zalex1953 Tested
+	  by: mnicholson, zalex1953
+
+2009-12-04 21:54 +0000 [r233116-233279]  David Vossel <dvossel at digium.com>
+
+	* configs/iax.conf.sample: clarify requirecalltoken option in
+	  iax.sample.conf (closes issue #16223) Reported by: bklang
+	  Patches: clarify-iax-requirecalltoken.patch uploaded by bklang
+	  (license 919)
+
+	* apps/app_voicemail.c: document and rename strip_control() in
+	  app_voicemail (closes issue #16291) Reported by: wdoekes
+
+2009-12-04 17:12 +0000 [r233092]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: Only do frame payload check for HOLD frames. This
+	  code was added for helping to debug the source of invalid HOLD
+	  frames. However, a side effect of this is that it will
+	  incorrectly report errors for frames that have an integer
+	  payload. Make the check for this block specific to the HOLD frame
+	  case.
+
+2009-12-04 16:59 +0000 [r233014-233091]  Matthias Nick <mnick at digium.com>
+
+	* pbx/pbx_config.c: Parse global variables or expressions in hint
+	  extensions Parse global variables or expressions in hint
+	  extensions. Like: exten => 400,hint,DAHDI/i2/${GLOBAL(var)}
+	  (closes issue #16166) Reported by: rmudgett Tested by: mnick,
+	  rmudgett
+
+	* main/dsp.c: Warning message gets displayed only once Added
+	  additional field 'int display_inband_dtmf_warning', which when
+	  set to '1' displays the warning ('Inband DTMF is not supported on
+	  codec %s. Use RFC2833'), and when set to '0' doesn't display the
+	  warning. Otherwise you would get hundreds of warnings every
+	  second. (closes issue #15769) Reported by: falves11 Patches:
+	  patch_15769_14.txt uploaded by mnick (license 874) Tested by:
+	  mnick, falves11
+
+2009-12-03 20:10 +0000 [r232820]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Deprecate "cz" in favor of "cs". Also,
+	  change the use of language codes so that language registers as a
+	  prefix, rather than an exact match. (closes issue #16272)
+	  Reported by: patrol-cz Patches: 20091203__issue16272.diff.txt
+	  uploaded by tilghman (license 14)
+
+2009-12-02 21:57 +0000 [r232581]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/manager.c: Send ack (response/message) after receiving
+	  manager action userevent (closes issue #16264) Reported by: dimas
+	  Patches: event-ack.patch uploaded by dimas (license 88)
+
+2009-12-02 19:03 +0000 [r232444]  David Vossel <dvossel at digium.com>
+
+	* apps/app_queue.c: fixes app_queue ao2 error (closes issue #16369)
+	  Reported by: vrban Patches: queue_issue_1.4.diff uploaded by
+	  dvossel (license 671) Tested by: dvossel
+
+2009-12-02 17:04 +0000 [r232355]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_amd.c: Fix a bug where if you hung up very quickly after
+	  calling AMD it would overwrite the AMDSTATUS of HANGUP with
+	  TOOLONG. (closes issue #16239) Reported by: CGMChris
+
+2009-12-02 16:59 +0000 [r232268-232350]  David Vossel <dvossel at digium.com>
+
+	* main/acl.c: ast_outaddrfor doesn't do htons() on port, looks odd
+	  in strace. (closes issue #16290) Reported by: wdoekes
+
+	* funcs/func_groupcount.c: fixes segfault in func_groupcount closes
+	  issue #16337) Reported by: Parantido Patches: issue_16337.diff
+	  uploaded by dvossel (license 671) Tested by: Parantido, dvossel
+
+2009-12-02 04:05 +0000 [r232165]  Terry Wilson <twilson at digium.com>
+
+	* main/channel.c: Fix compiling without devmode (closes issue
+	  #16367) Reported by: falves11
+
+2009-12-02 00:42 +0000 [r232090]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Do not modify the gain settings on data
+	  calls. (The digital flag actually represents a data call.)
+	  (closes issue #15972) Reported by: udosw Patches:
+	  transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
+	  Tested by: alecdavis
+
+2009-12-01 23:25 +0000 [r232007]  Russell Bryant <russell at digium.com>
+
+	* main/file.c: Fix a warning pointed out by buildbot.
+
+2009-12-01 21:52 +0000 [r231911-231926]  Jeff Peeler <jpeeler at digium.com>
+
+	* main/channel.c: log channel name in dev mode as well
+
+	* main/channel.c: Fix crash with invalid frame data The crash was
+	  happening as a result of a frame containing an invalid data
+	  pointer, but was set with data length of zero. The few times the
+	  issue was reproduced it _seemed_ that the frame was queued
+	  properly, that is the data pointer was set to NULL. I never could
+	  reproduce the crash so as a last resort the crash has been fixed,
+	  but a check in __ast_read has been added to give as much
+	  information about the source of problematic frames in the future.
+	  (closes issue #16058) Reported by: atis
+
+2009-12-01 21:14 +0000 [r231853]  David Vossel <dvossel at digium.com>
+
+	* main/pbx.c: WaitExten m option with no parameters generates frame
+	  with zero datalen but non-null data ptr
+
+2009-12-01 15:34 +0000 [r231614-231740]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/file.c: Ignore unknown formats in ast_format_str_reduce()
+	  and return an error if no know formats are found.
+
+	* apps/app_voicemail.c, include/asterisk/file.h, main/file.c,
+	  main/app.c: Remove duplicate entries from voicemail format lists.
+	  This prevents app_voicemail from entering an infinite loop when
+	  the same format is specified twice in the format list. (closes
+	  issue #15625) Reported by: Shagg63 Tested by: mnicholson Review:
+	  https://reviewboard.asterisk.org/r/429/
+
+2009-11-30 17:14 +0000 [r231437-231441]  David Vossel <dvossel at digium.com>
+
+	* main/rtp.c: fixes crash caused by RTP comfort noise payload
+	  greater than 24 bytes AST-2009-010 (closes issue #16242) Reported
+	  by: amorsen Patches: issue16242.diff uploaded by oej (license
+	  306) Tested by: amorsen, oej, dvossel
+
+	* apps/app_queue.c: app_queue crashes randomly, often during
+	  call-transfers In app_queue, it is possible for a call_queue to
+	  be destroyed while another object still holds a pointer to it.
+	  This patch converts call_queue objects to ao2 objects allowing
+	  them to be ref counted. This makes it safe for the queue_ent
+	  object in queue_exec() to reference it's parent call_queue even
+	  after it has left the queue. (closes issue #15686) Reported by:
+	  Hatrix Patches: v2_queue_ao2.diff uploaded by dvossel (license
+	  671) Tested by: dvossel, aragon Review:
+	  https://reviewboard.asterisk.org/r/427/
+
+2009-11-25 22:31 +0000 [r231298]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c: After a frame duplication failure, unlock the
+	  channel before returning.
+
+2009-11-25 21:38 +0000 [r231233-231235]  David Vossel <dvossel at digium.com>
+
+	* apps/app_dial.c: fixes solaris segfault on dial with verbosity >=
+	  3 (closes issue #16193) Reported by: asgaroth Patches:
+	  bug_16193_1.4.21.2_vers.diff uploaded by snuffy (license 35)
+	  Tested by: asgaroth, snuffy
+
+	* channels/chan_sip.c: fixes conditional jump or move depending on
+	  uninitialised STACK value (closes issue #16261) Reported by:
+	  edguy3 Patches: edguy16261.patch uploaded by edguy3 (license 917)
+
+2009-11-23 15:31 +0000 [r230772-230875]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: When 'sip set debug' is enabled, and the
+	  last line of an incoming SIP message is not properly newline
+	  terminated, ensure that that line is included in the debug
+	  output. (part of issue #16268)
+
+	* main/editline/makelist.in, channels/chan_sip.c,
+	  channels/ring_tone.h, channels/busy_tone.h: Correct fix for issue
+	  #16268... the reporter's original patch was very close to
+	  correct.
+
+	* channels/chan_sip.c: Ensure that SDP parsing does not ignore the
+	  last line of the SDP. (closes issue #16268) Reported by: sgimeno
+
+2009-11-20 20:53 +0000 [r230627]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_features.c: Copy the peer CDR's userfield to the bridge
+	  CDR if it exists. This is necessary for the recordagentcalls
+	  option in chan_agent to store the recorded file name in the
+	  bridge CDR. (closes issue #14590) Reported by: msetim Patches:
+	  queue_agent_userfield.patch uploaded by Laureano (license 265)
+	  Tested by: Laureano, mnicholson
+
+2009-11-19 21:22 +0000 [r230508]  David Vossel <dvossel at digium.com>
+
+	* apps/app_mixmonitor.c: fixes MixMonitor thread not exiting when
+	  StopMixMonitor is used (closes issue #16152) Reported by: AlexMS
+	  Patches: stopmixmonitor_1.4.diff uploaded by dvossel (license
+	  671) Tested by: dvossel, AlexMS Review:
+	  https://reviewboard.asterisk.org/r/424/
+
+2009-11-19 16:09 +0000 [r230469]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/asterisk.c: Update copyright year in visible output. (cli)
+	  Spotted by Stuart Henderson
+
+2009-11-30  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27.1
+
+	* AST-2009-010
+
+	* SDP parser regression fix (issue #16268)
+
+2009-11-18  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27
+
+2009-11-13  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc5
+
+2009-11-12 16:41 +0000 [r229669]  David Vossel <dvossel at digium.com>
+
+	* funcs/func_audiohookinherit.c: fixes merging error, datastore was
+	  being freed in the wrong function. (closes issue #16219) Reported
+	  by: aragon
+
+2009-11-11 19:46 +0000 [r229498]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c: Solaris doesn't like NULL going to ast_log Solaris
+	  will crash if NULL is passed to ast_log. This simple patch simply
+	  uses S_OR to get around this. (closes issue #15392) Reported by:
+	  yrashk
+
+2009-11-10 22:09 +0000 [r229360]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c: If two pattern classes start with the same digit and
+	  have the same number of characters, they will compare equal. The
+	  example given in the issue report is that of [234] and [246],
+	  which have these characteristics, yet they are clearly not
+	  equivalent. The code still uses these two characteristics, yet
+	  when the two scores compare equal, an additional check will be
+	  done to compare all characters within the class to verify
+	  equality. (closes issue #15421) Reported by: jsmith Patches:
+	  20091109__issue15421__2.diff.txt uploaded by tilghman (license
+	  14) Tested by: jsmith, thedavidfactor
+
+2009-11-10 21:45 +0000 [r229355]  David Ruggles <thedavidfactor at gmail.com>
+
+	* doc/externalivr.txt: Fix ExternalIVR Documentation Remove
+	  documentation for event that doesn't function (closes issue
+	  #16220) Reported by: thedavidfactor Patches:
+	  externalivr.txt.20091110.1622.patch uploaded by thedavidfactor
+	  (license 903)
+
+2009-11-10 20:03 +0000 [r229281]  Joshua Colp <jcolp at digium.com>
+
+	* codecs/codec_g726.c: Remove broken support for direct transcoding
+	  between G.726 RFC3551 and G.726 AAL2. On some systems the
+	  translation core would actually consider g726aal2 -> g726 ->
+	  signed linear to be a quicker path then g726aal2 -> signed linear
+	  which exposed this problem. (closes issue #15504) Reported by:
+	  globalnetinc
+
+2009-11-10 17:23 +0000 [r229191]  David Ruggles <thedavidfactor at gmail.com>
+
+	* doc/externalivr.txt: Document ExternalIVR event tag collision
+	  ExternalIVR uses the D tag for two different event types. This
+	  documents that behavior and how to differentiate between the two
+	  cases. Also includes a minor spelling fix and clarification
+	  (closes issue #16211) Reported by: thedavidfactor Patches:
+	  externalivr.txt.20091109.1507.patch uploaded by thedavidfactor
+	  (license 903)
+
+2009-11-10 17:15 +0000 [r229167]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: don't crash on log message in solaris
+	  AST-2009-006 (closes issue #16206) Reported by: bklang Tested by:
+	  bklang
+
+2009-11-10 15:22 +0000 [r229091]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Reverted revision 202022. (closes issue
+	  #16175) Reported by: paul-tg
+
+2009-11-09  Leif Madsen <lmadsen at digium.com>
+
+	* Release Astersik 1.4.27-rc4
+
+2009-11-09 15:37 +0000 [r228896]  Leif Madsen <lmadsen at digium.com>
+
+	* main/channel.c: Update WARNING message. Update a WARNING message
+	  to give a suggested fix when encountered. (closes issue #16198)
+	  Reported by: atis Tested by: atis
+
+2009-11-09 14:16 +0000 [r228827]  Matthew Nicholson <mnicholson at digium.com>
+
+	* include/asterisk/lock.h: Perform limited bounds checking when
+	  destroying ast_mutex_t structures to make sure we don't try to
+	  use negative indices. (closes issue #15588) Reported by: zerohalo
+	  Patches: 20090820__issue15588.diff.txt uploaded by tilghman
+	  (license 14) Tested by: zerohalo
+
+2009-11-06 22:33 +0000 [r228692]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c: fixes audiohook write crash occuring in chan_spy
+	  whisper mode. After writing to the audiohook list in ast_write(),
+	  frames were being freed incorrectly. Under certain conditions
+	  this resulted in a double free crash. (closes issue #16133)
+	  Reported by: wetwired (closes issue #16045) Reported by:
+	  bluecrow76 Patches: issue16045.diff uploaded by dvossel (license
+	  671) Tested by: bluecrow76, dvossel, habile
+
+2009-11-06 18:32 +0000 [r228547]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Don't overwrite caller ID name on a trunk
+	  with the configured fullname when using users.conf (issue
+	  ABE-1989)
+
+2009-11-06  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc3
+
+2009-11-06 17:07 +0000 [r228418]  David Vossel <dvossel at digium.com>
+
+	* codecs/codec_ilbc.c: fixes segfault in iLBC For reasons not yet
+	  known, it appears possible for an ast_frame to have a datalen
+	  greater than zero while the actual data is NULL during Packet
+	  Loss Concealment. Most codecs don't support PLC so this doesn't
+	  affect them. This patch catches the malformed frame and prevents
+	  the crash from occuring. Additional efforts to determine why it
+	  is possible for a frame to look like this are still being
+	  investigated. (issue #16979)
+
+2009-11-06 16:41 +0000 [r228409]  Joshua Colp <jcolp at digium.com>
+
+	* main/abstract_jb.c: Fix a bug caused by a partially invalid frame
+	  (from the jitterbuffer) passing through the Asterisk core.
+	  (closes issue #15560) Reported by: jvandal (closes issue #15709)
+	  Reported by: covici
+
+2009-11-06 16:26 +0000 [r228378]  Matthew Nicholson <mnicholson at digium.com>
+
+	* funcs/func_base64.c, main/utils.c: Properly handle '=' while
+	  decoding base64 messages and null terminate strings returned from
+	  BASE64_DECODE. (closes issue #15271) Reported by: chappell
+	  Patches: base64_fix.patch uploaded by chappell (license 8) Tested
+	  by: kobaz
+
+2009-11-06 15:41 +0000 [r228272-228338]  David Vossel <dvossel at digium.com>
+
+	* main/astfd.c: fixes crash in astfd.c (closes issue #15981)
+	  Reported by: slavon
+
+	* funcs/func_audiohookinherit.c: fixes memory leak in
+	  func_audiohookinherit.c (closes issue 0015394) Reported by:
+	  boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks
+	  (license 790) Tested by: dbrooks, boroda
+
+2009-11-05 19:14 +0000 [r228079]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_vpb.cc: Fix crash on VPB exception when no hardware
+	  is present. (closes issue #14970) Reported by: tzafrir Patches:
+	  vpb_exception.diff uploaded by tzafrir (license 46) Tested by:
+	  markwaters
+
+2009-11-05 18:59 +0000 [r228078]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_misdn.c: chan_misdn Asterisk 1.4.27-rc2 crash Crash
+	  related to chan_misdn connection. Patch submitted by
+	  gknispel_proformatique, tested by francesco_r. "I have many crash
+	  since i have upgraded to Asterisk 1.4.27-rc2. Attached a full
+	  bt." This patch zeros out an ast_frame. (closes issue #16041)
+	  Reported by: francesco_r
+
+2009-11-04 23:47 +0000 [r227944]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_monitor.c: Fix incorrect filename comparsion after
+	  monitor file change The logic to detect if a requested file is
+	  indeed a different file from the current file was incorrect. The
+	  main issue being confusion of the use of filename_base which was
+	  previously set without pathing information and then compared to
+	  another full path. Robust file comparison logic has been added to
+	  properly check if two files are the same even if symlinks are
+	  used. (closes issue #15313) Reported by: caspy Patches:
+	  20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license
+	  325) but mostly tilghman's work
+
+2009-11-04 20:52 +0000 [r227758-227827]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_dial.c: This patch modifies the Dial application to
+	  monitor the calling channel for hangups while playing back
+	  announcements. (closes issue #16005) Reported by: falves11
+	  Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson
+	  (license 96) Tested by: mnicholson, falves11 Review:
+	  https://reviewboard.asterisk.org/r/407/
+
+	* channels/chan_sip.c: Modify the SDP parsing code to parse session
+	  and media level items separately. With the new code, media level
+	  proprieties should no longer be confused with session level
+	  proprieties. This change also reorganizes some of the SDP parsing
+	  code which should make it easier to manage in the future. (closes
+	  issue #14994) Reported by: frawd Tested by: frawd, mnicholson,
+	  file Review: https://reviewboard.asterisk.org/r/385/
+
+2009-11-04 19:25 +0000 [r227700-227735]  Joshua Colp <jcolp at digium.com>
+
+	* static-http/prototype.js: Fix a security issue where it may be
+	  possible for someone to execute a cross-site AJAX request
+	  exploit. (AST-2009-009)
+
+	* channels/chan_sip.c: Fix a security issue where sending a
+	  REGISTER with a differing username in the From URI and
+	  Authorization header would reveal whether it was valid or not.
+	  (AST-2009-008)
+
+2009-11-03 17:55 +0000 [r227275]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Make sure the outgoing flag is cleared if
+	  a new channel fails to get created for outgoing calls. This is
+	  the relevant portion of asterisk/trunk -r226648
+
+2009-11-03 15:36 +0000 [r227166]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Fix a bug where an RPID header could be
+	  generated with a blank username in the URI. (closes issue #15909)
+	  Reported by: kobaz
+
+2009-11-03 10:48 +0000 [r227088-227090]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Fixing bug before someone reports it...
+
+	* channels/chan_sip.c: Adding IP address in Contact ACL log message
+	  and removing redundant message (based on kpfleming's feedback)
+
+	* channels/chan_sip.c: Use proper response code when violating
+	  Contact ACL's. Review: https://reviewboard.asterisk.org/r/415/
+	  Thanks kpfleming for a quick review. (EDVX-003)
+
+2009-11-02 20:52 +0000 [r226972]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_sip.c: SIP channel name uniqueness SIP channel
+	  names were supposed to be unique by way of a name suffix derived
+	  from the pointer to the channel's private data. Uniqueness was
+	  preserved on 32-bit systems, but not on 64-bit systems. This
+	  patch, as suggested by kpfleming, replaces this suffix with a
+	  simple incremented unsigned int. (closes issue #15152) Reported
+	  by: palbrecht Review: https://reviewboard.asterisk.org/r/420/
+
+2009-11-02 18:08 +0000 [r226889]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Fix a bug where the recorded privacy
+	  introduction file would not get removed if the caller hung up
+	  while the called party had not yet answered. This was fixed by
+	  introducing an argument to the 'n' option which, when enabled,
+	  removes the introduction file under all scenarios. This was done
+	  to preserve the behavior that has existed for quite some time.
+	  (closes issue #14674) Reported by: ulogic Patches: bug14674.patch
+	  uploaded by jpeeler (license 325)
+
+2009-11-02 17:14 +0000 [r226811]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/rc.redhat.asterisk: Don't allow two separate
+	  instances of safe_asterisk when restarting from the init script.
+	  (closes issue #14562) Reported by: davidw Patches: Initially
+	  20091022__issue14562.diff.txt uploaded by tilghman (license 14)
+	  Modified to 20091030__Issue14562_diff.txt uploaded by davidw
+	  (license 780) Tested by: davidw
+
+2009-11-02 15:31 +0000 [r226688-226736]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: fixes crash on iterator_destroy on
+	  uninitialized iterator (closes issue #16162) Reported by: krn
+
+	* channels/chan_iax2.c: changes calltoken debug messages from
+	  LOG_NOTICE to LOG_DEBUG like they are supposed to be (closes
+	  issue #16144) Reported by: aragon
+
+2009-10-29 18:11 +0000 [r226531]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_local.c, doc/localchannel.txt: Add an option to
+	  enabling passing music on hold start and stop requests through
+	  instead of acting on them in chan_local. (closes issue #14709)
+	  Reported by: dimas
+
+2009-10-28 20:06 +0000 [r226377-226382]  Leif Madsen <lmadsen at digium.com>
+
+	* configs/sip.conf.sample: Update documentation in sip.conf.sample.
+	  Update the documentation in sip.conf.sample in order to make it
+	  more clear that directmedia/canreinvite do not cause Asterisk to
+	  ignore reINVITEs. It is only used to stop Asterisk from
+	  generating a reINVITE, but does not stop it from accepting them
+	  if necessary. (closes issue #15644) Reported by: lmadsen
+
+	* doc/channelvariables.txt: Update CALLINGSUBADDR channel variable
+	  documentation. (closes issue #15734) Reported by: alecdavis
+	  Patches: channelvariables.tex.diff.txt uploaded by alecdavis
+	  (license 585) Tested by: alecdavis
+
+2009-10-28 18:02 +0000 [r226138-226304]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk/linkedlists.h: Fix documentation (pointed out by
+	  TheDavidFactor on #-dev)
+
+	* main/manager.c: Manager output is not always NULL-terminated, so
+	  force a NULL at the end of the filestream. (closes issue #15495)
+	  Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded
+	  by tilghman (license 14) Tested by: pdf
+
+2009-10-26 22:13 +0000 [r225957]  Tzafrir Cohen <tzafrir.cohen at xorcom.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac: detect
+	  ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabi * Set
+	  OSARCH to linux-gnu even if host_os is linux-gnueabi * When
+	  checking if we are Linux, check OSARCH rather than host_os The
+	  newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather
+	  than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even
+	  in such a case. OSARCH is tested for the value of 'linux-gnu' in
+	  one or two places in the tree. This patch also fixes the check
+	  libcap to check for $OSARCH rather than $host_os . See also:
+	  http://wiki.debian.org/ArmEabiPort
+
+2009-10-23 14:00 +0000 [r225581]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile: Don't force menuselect.makeopts to be rebuilt on every
+	  build. For some reason the menuselect.makeopts file was listed as
+	  PHONY in the Makefile, resulting in 'make' needing to rebuild it
+	  for every build. This then resulted in the embedded module rules
+	  being rebuilt on every build, which can be slow and is
+	  unnecessary. This patch fixes the problem by properly allowing
+	  'make' to know when the menuselect.makeopts file needs to be
+	  rebuilt (defining the proper dependencies).
+
+2009-10-22 21:51 +0000 [r225484]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/valgrind.txt, contrib/valgrind.supp (added): Clean valgrind
+	  output by suppressing false errors. Update valgrind.txt
+	  documentation and add valgrind.supp file in order to allow those
+	  who are creating valgrind output to have less false errors in the
+	  logfile. (closes issue #16007) Reported by: atis Patches:
+	  valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp
+	  uploaded by atis (license 242) Tested by: atis, amorsen
+
+2009-10-21 20:58 +0000 [r225243]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: IAX2: VNAK loop caused by signaling frames
+	  with no destination call number It is possible for the PBX thread
+	  to queue up signaling frames before a destination call number is
+	  received. This can result in signaling frames being sent out with
+	  no destination call number. Since recent versions of Asterisk
+	  require accurate destination callnumbers for all Full Frames,
+	  this can cause a VNAK loop to occur. To resolve this no signaling
+	  frames are sent until a destination callnumber is received, and
+	  destination call numbers are now only required for iax_pvt
+	  matching when the frame is an ACK. Review:
+	  https://reviewboard.asterisk.org/r/413/
+
+2009-10-21 16:44 +0000 [r225169-225171]  Russell Bryant <russell at digium.com>
+
+	* main/translate.c: Revert 225169, as this doesn't account for the
+	  possibility of a list of frames.
+
+	* main/translate.c: Isolate the frame returned from
+	  ast_translate().
+
+2009-10-21 16:02 +0000 [r225103-225105]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/pbx.c, apps/app_meetme.c, include/asterisk/channel.h: Fix
+	  documentation for ast_softhangup() and correct the misuse
+	  thereof. (closes issue #16103) Reported by: majorbloodnok
+
+	* apps/app_voicemail.c: Suffix is not needed for a match
+
+2009-10-21 14:37 +0000 [r225032]  David Vossel <dvossel at digium.com>
+
+	* configs/iax.conf.sample, channels/chan_sip.c,
+	  configs/sip.conf.sample, channels/chan_iax2.c: IAX/SIP
+	  shrinkcallerid option The shrinking of caller id removes '(', '
+	  ', ')', non-trailing '.', and '-' from the string. This means
+	  values such as 555.5555 and test-test result in 555555 and
+	  testtest. There are instances, such as Skype integration, where a
+	  specific value is passed via caller id that must be preserved
+	  unmodified. This patch makes the shrinking of caller id optional
+	  in chan_sip and chan_iax in order to support such cases. By
+	  default this option is on to preserve previous expected behavior.
+	  (closes issue #15940) Reported by: dimas Patches: v2-15940.patch
+	  uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c
+	  uploaded by dvossel (license 671) Tested by: dvossel Review:
+	  https://reviewboard.asterisk.org/r/408/
+
+2009-10-21 02:59 +0000 [r224931]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/translate.h, main/dsp.c, main/frame.c,
+	  main/translate.c, include/asterisk/dsp.h, codecs/codec_dahdi.c,
+	  include/asterisk/frame.h: Isolate frames returned from a DSP
+	  instance or codec translator. The reasoning for these changes are
+	  the same as what I wrote in the commit message for rev 222878.
+
+2009-10-20 22:07 +0000 [r224855]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/audiohook.c: Pay attention to the return value of the
+	  manipulate function. While this looks like an optimization, it
+	  prevents a crash from occurring when used with certain audiohook
+	  callbacks (diagnosed with SVN trunk, backported to 1.4 to keep
+	  the source consistent across versions).
+
+2009-10-20 17:46 +0000 [r224773]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: Add support for relaying early media in the
+	  features attended transfer option. (closes issue #14828) Reported
+	  by: licedey
+
+2009-10-19 23:44 +0000 [r224670]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/rtp.c: Correct timestamp calculations when RTP sample rates
+	  over 8kHz are used. While testing some endpoints that support
+	  16kHz and 32kHz sample rates, some log messages were generated
+	  due to calc_rxstamp() computing timestamps in a way that produced
+	  odd results, so this patch sanitizes the result of the
+	  computations.
+
+2009-10-19 19:47 +0000 [r224565]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_dial.c: Do not attempt early media bridging (ie: direct
+	  RTP setup) if options are enabled that should prevent it. (closes
+	  issue #14763) Reported by: cupotka
+
+2009-10-17 01:32 +0000 [r224330]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Fix stale caller id data from being
+	  reported in AMI NewChannel event The problem here is that
+	  chan_dahdi is designed in such a way to set certain values in the
+	  dahdi_pvt only once. One of those such values is the configured
+	  caller id data in chan_dahdi.conf. For PRI, the configured caller
+	  id data could be overwritten during a call. Instead of saving the
+	  data and restoring, it was decided that for all non-analog
+	  channels it was simply best to not set the configured caller id
+	  in the first place and also clear it at the end of the call.
+	  (closes issue #15883) Reported by: jsmith
+
+2009-10-16 20:25 +0000 [r224260]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Never released PRI channels when using
+	  Busy() or Congestion() dialplan apps. When the Busy() or
+	  Congestion() application is used towards ISDN (an ISDN progress
+	  is sent), the responding ISDN Disconnect or Release may contain
+	  the ISDN cause user busy or one of the congestion causes. In
+	  chan_dahdi.c these causes will only set the needbusy or
+	  needcongestion flags and not activate the softhangup procedure.
+	  Unfortunately only the latter can interrupt the endless wait loop
+	  of Busy()/Congestion(). Result: PRI channels staying in state
+	  busy for the rest of asterisk life or until the other end times
+	  out and forces the call to clear. (in issue 0014292) Reported by:
+	  tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso
+	  (license 564) (This patch is unrelated to the issue.)
+
+2009-10-13 20:58 +0000 [r223955]  Jean Galarneau <jgalarneau at digium.com>
+
+	* channels/chan_dahdi.c: Fix PRI timer T309 operation
+
+2009-10-12 23:12 +0000 [r223804]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_dial.c: Ensure ringing continues for branched calls
+	  after progress is received While waiting for an answer, don't
+	  send progress for branched calls for which ringing was sent.
+	  (closes issue #15028) Reported by: fnordian
+
+2009-10-12 15:30 +0000 [r223692]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: Remove automatic switching from T.38 to
+	  voice mode in chan_sip. chan_sip has some code to automatically
+	  switch from T.38 mode to voice mode when a voice frame is written
+	  to the channel while it is in T.38 mode; this was intended to
+	  handle the situation when a FAX transmission has ended and the
+	  channel is not yet hung up, but is causing problems at the
+	  beginning of FAX sessions as well when there are still voice
+	  frames 'in flight' at the time the T.38 negotiation completes.
+	  This patch removes the automatic switchover. (issue #16025)
+	  Reported by: jamicque
+
+2009-10-11 18:34 +0000 [r223485-223550]  Russell Bryant <russell at digium.com>
+
+	* apps/app_queue.c: Remove a duplicate ao2_iterator_destroy().
+
+	* main/autoservice.c: Remove some unnecessary code.
+
+	* main/autoservice.c: Don't use data outside of its scope. The
+	  purpose of this code was to have a hangup frame put on the list
+	  of deferred frames. However, the code that read the hangup frame
+	  was outside of the scope of where the hangup frame was declared.
+
+2009-10-09 18:20 +0000 [r223225]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c: Signal timeouts by returning AST_CONTROL_RINGING
+	  when originating calls. (closes issue #15104) Reported by:
+	  nblasgen Patches: manager-timeout1.diff uploaded by mnicholson
+	  (license 96) Tested by: nblasgen, mnicholson
+
+2009-10-09 18:17 +0000 [r223213]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_dial.c: Fix potential memory leak in app_dial.c
+
+2009-10-09 17:52 +0000 [r223142-223205]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: fixes sip registration using authuser in
+	  user.conf (closes issue #14954) Reported by: tornblad Tested by:
+	  mmichelson, tornblad, dvossel
+
+	* channels/chan_sip.c: 'auth=' did not parse md5 secret correctly
+	  (closes issue https://issues.asterisk.org/view.php?id=15949)
+	  Reported by: ebroad Patches: authparsefix.patch uploaded by
+	  ebroad (license 878) 15949_trunk.diff uploaded by dvossel
+	  (license 671) Tested by: ebroad
+
+2009-10-08 19:45 +0000 [r222878]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/file.h, main/frame.c, main/file.c,
+	  include/asterisk/frame.h: Make filestream frame handling safer by
+	  isolating frames before returning them. This patch is related to
+	  a number of issues on the bug tracker that show crashes related
+	  to freeing frames that came from a filestream. A number of fixes
+	  have been made over time while trying to figure out these
+	  problems, but there re still people seeing the crash. (Note that
+	  some of these bug reports include information about other
+	  problems. I am specifically addressing the filestream frame crash
+	  here.) I'm still not clear on what the exact problem is. However,
+	  what is _very_ clear is that we have seen quite a few problems
+	  over time related to unexpected behavior when we try to use
+	  embedded frames as an optimization. In some cases, this
+	  optimization doesn't really provide much due to improvements made
+	  in other areas. In this case, the patch modifies filestream
+	  handling such that the embedded frame will not be returned.
+	  ast_frisolate() is used to ensure that we end up with a
+	  completely mallocd frame. In reality, though, we will not
+	  actually have to malloc every time. For filestreams, the frame
+	  will almost always be allocated and freed in the same thread.
+	  That means that the thread local frame cache will be used. So,
+	  going this route doesn't hurt. With this patch in place, some
+	  people have reported success in not seeing the crash anymore.
+	  (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon
+	  Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell
+	  (license 2) Tested by: aragon, russell (closes issue #15817)
+	  Reported by: zerohalo Tested by: zerohalo (closes issue #15845)
+	  Reported by: marhbere Review:
+	  https://reviewboard.asterisk.org/r/386/
+
+2009-10-08 19:45 +0000 [r222877]  David Vossel <dvossel at digium.com>
+
+	* main/netsock.c, include/asterisk/netsock.h: fixes an
+	  ast_netsock_list memory leak. ABE-1998 Review:
+	  https://reviewboard.asterisk.org/r/395/
+
+2009-10-08 16:33 +0000 [r222691-222797]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn_config.c: Fix memory leak if chan_misdn config
+	  parameter is repeated. Memory leak when the same config option is
+	  set more than once in an misdn.conf section. Why must this be
+	  considered? Templates! Defining a template with default port
+	  options and later adding to or overriding some of them. Patches:
+	  memleak-misdn.patch JIRA ABE-1998
+
+	* channels/chan_misdn.c: chan_misdn.c:process_ast_dsp() memory leak
+	  misdn.conf: astdtmf must be set to "yes". With "no", buffer loss

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