[asterisk-commits] lmadsen: tag 1.4.27-rc5 r229963 - /tags/1.4.27-rc5/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Nov 13 11:04:01 CST 2009


Author: lmadsen
Date: Fri Nov 13 11:03:57 2009
New Revision: 229963

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=229963
Log:
Importing files for 1.4.27-rc5 release.

Added:
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    tags/1.4.27-rc5/.version   (with props)
    tags/1.4.27-rc5/ChangeLog   (with props)

Added: tags/1.4.27-rc5/.lastclean
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--- tags/1.4.27-rc5/ChangeLog (added)
+++ tags/1.4.27-rc5/ChangeLog Fri Nov 13 11:03:57 2009
@@ -1,0 +1,26809 @@
+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
+	  does not occur. The translated frame "f2" when passing through
+	  ast_dsp_process() is not freed whenever it is not used further in
+	  process_ast_dsp(). Then in the end it is never ever freed.
+	  Patches: translate.patch JIRA ABE-1993
+
+2009-10-07 17:41 +0000 [r222542]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: crash on transfer handle_invite_replaces()
+	  attempts to uplock a pvt's owner channel without first verifing
+	  that it exists. (issue #16027)
+
+2009-10-06 23:51 +0000 [r222393-222462]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Add missing unlock(s) in dahdi_read (two
+	  cases in trunk) (closes issue #15683) Reported by: alecdavis
+
+	* channels/chan_dahdi.c: Fix potential crash when entire span
+	  request is received. The variable index used in this scenario for
+	  accessing the dahdi_pvts was wrong and was most likely copied
+	  from the several other places it is used correctly. (closes issue
+	  #15998) Reported by: tsearle Patches: dahdi_reset_crash.patch
+	  uploaded by tsearle (license 373)
+
+2009-10-06  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc2
+
+2009-10-06 01:16 +0000 [r222152]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/astobj2.c, include/asterisk/astobj2.h,
+	  res/res_musiconhold.c, apps/app_queue.c, channels/chan_iax2.c:
+	  Fix ao2_iterator API to hold references to containers being
+	  iterated. See Mantis issue for details of what prompted this
+	  change. Additional notes: This patch changes the ao2_iterator API
+	  in two ways: F_AO2I_DONTLOCK has become an enum instead of a
+	  macro, with a name that fits our naming policy; also, it is now
+	  necessary to call ao2_iterator_destroy() on any iterator that has
+	  been created. Currently this only releases the reference to the
+	  container being iterated, but in the future this could also
+	  release other resources used by the iterator, if the iterator
+	  implementation changes to use additional resources. (closes issue
+	  #15987) Reported by: kpfleming Review:
+	  https://reviewboard.asterisk.org/r/383/
+
+2009-10-02 17:32 +0000 [r222026]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Removes unnecessary unlock, clarifies a
+	  memcpy.
+
+2009-10-02 16:58 +0000 [r221776-221970]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/astobj2.c: Ensure the result of the hash function is
+	  positive. Negative array offsets suck.
+
+	* main/asterisk.c, main/rtp.c, main/say.c: Fix a bunch of
+	  off-by-one errors
+
+2009-10-01 23:18 +0000 [r221769]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/misdn/isdn_lib_intern.h:
+	  Occasionally losing use of B channels in chan_misdn. I have not
+	  been able to reproduce the problem of losing channels. However, I
+	  have seen in the code a reentrancy problem that might give these
+	  symptoms. The reentrancy patch does several things: 1) Guards B
+	  channel and B channel structure allocation. 2) Makes the B
+	  channel structure find routines more precise in locating records.
+	  3) Never leave a B channel allocated if we received cause 44. The
+	  last item may cause temporary outgoing call problems, but they
+	  should clear when the line becomes idle. (closes issue #15490)
+	  Reported by: slutec18 Patches:
+	  issue15490_channel_alloc_reentrancy.patch uploaded by rmudgett
+	  (license 664) Tested by: rmudgett, slutec18 (closes issue #15458)
+	  Reported by: FabienToune Patches:
+	  issue15458_channel_alloc_reentrancy.patch uploaded by rmudgett
+	  (license 664) Tested by: FabienToune, rmudgett, slutec18
+
+2009-10-01 15:24 +0000 [r221360-221588]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Use unsigned ints for portinuri flags.
+
+	* channels/chan_sip.c: Make portinuri a bitfield.
+
+	* channels/chan_sip.c, configs/sip.conf.sample: Fix SRV lookup and
+	  Request-URI generation in chan_sip. This patch adds a new field
+	  "portinuri" to the sip dialog struct and the sip peer struct.
+	  That field is used during RURI generation to determine if the
+	  port should be included in the RURI. It is also used in some
+	  places to determine if an SRV lookup should occur. (closes issue
+	  #14418) Reported by: klaus3000 Tested by: klaus3000, mnicholson
+	  Review: https://reviewboard.asterisk.org/r/369/
+
+2009-09-30 19:02 +0000 [r221303]  Matthias Nick <mnick at digium.com>
+
+	* funcs/func_strings.c: changed the prototype definition of
+	  csv_quote
+
+2009-09-30 16:55 +0000 [r221200]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c: Avoid a potential NULL dereference. (closes issue
+	  #15865) Reported by: kobaz Patches: 20090915__issue15865.diff.txt
+	  uploaded by tilghman (license 14) Tested by: kobaz
+
+2009-09-30 15:41 +0000 [r221153-221157]  Matthias Nick <mnick at digium.com>
+
+	* configs/cdr_custom.conf.sample, funcs/func_strings.c: added a new
+	  dialplan function 'CSV_QUOTE' and changed the
+	  cdr_custom.sample.conf (closes issue #15471) Reported by: dkerr
+	  Patches: csv_quote_14.txt uploaded by mnick (license ) Tested by:
+	  mnick
+
+	* funcs/func_strings.c: check bounds - prevents for buffer overflow
+
+2009-09-30 14:49 +0000 [r221086]  Terry Wilson <twilson at digium.com>
+
+	* main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h,
+	  configs/sip.conf.sample: Change the SSRC by default when our
+	  media stream changes Be default, change SSRC when doing an audio
+	  stream changes Asterisk doesn't honor marker bit when reinvited
+	  to already-bridged RTP streams,resulting in far-end stack
+	  discarding packets with "old" timestamps that areactually part of
+	  a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever
+	  there is a reinvite, unless the 'constantssrc' is set to true in
+	  sip.conf. The original issue reported to Digium support detailed
+	  the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based
+	  Application Server Call comes in fromITSP, Asterisk dials the app
+	  server which sends a re-invite back toAsterisk--not to negotiate
+	  to send media directly to the ITSP, but to indicatethat it's
+	  changing the stream it's sending to Asterisk. The app
+	  servergenerates a new SSRC, sequence numbers, timestamps, and
+	  sets the marker bit on the new stream. Asterisk passes through
+	  the teimstamp of the new stream, butdoes not reset the SSRC,
+	  sequence numbers, or set the marker bit. When the timestamp on
+	  the new stream is older than the timestamp on the originalstream,
+	  the ITSP (which doesn't know there has been any change) discards
+	  the newframes because it thinks they are too old. This patch
+	  addresses this by changing the SSRC on a stream update unless
+	  constantssrc=true is set in sip.conf. Review:
+	  https://reviewboard.asterisk.org/r/374/
+
+2009-09-29 20:14 +0000 [r220907]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_chanspy.c: Avoid a deadlock in chanspy, just in case the
+	  spyee is masqueraded and chanspy_ds_chan_fixup() is called with
+	  the channel locked. (closes issue #15965) Reported by: atis
+	  Patches: chanspy-deadlock-fix1.diff uploaded by mnicholson
+	  (license 96) Tested by: atis
+
+2009-09-29 17:59 +0000 [r220873]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Reduce CPU usage related to building a peer
+	  merely for devicestates. This fixes a 100% CPU problem in the SIP
+	  driver, found by profiling the driver while the problem was
+	  occurring. (closes issue #14309) Reported by: pkempgen Patches:
+	  20090924__issue14309.diff.txt uploaded by tilghman (license 14)
+	  Tested by: pkempgen, vrban
+
+2009-09-28 19:09 +0000 [r220717]  Sean Bright <sean at malleable.com>
+
+	* Makefile.rules: When selecting DONT_OPTIMIZE in menuselect,
+	  explicitly pass -O0 to the compiler so we override any default
+	  optimization levels for a particular install.
+
+2009-09-24 19:39 +0000 [r220288]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_playback.c, main/pbx.c, apps/app_disa.c: Implicitly
+	  sending a progress signal breaks some applications. Call
+	  Progress() in your dialplan if you explicitly want progress to be
+	  sent. (Reverts change 216430, closes issue #15957) Reported by:
+	  Pavel Troller on the Asterisk-Dev mailing list
+	  http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
+
+2009-09-24 18:18 +0000 [r220099-220213]  Sean Bright <sean at malleable.com>
+
+	* Makefile: Resolve parallel build warnings. Reported by Klaus
+	  Darilion on the asterisk-dev mailing list.
+
+	* Makefile, build_tools/mkpkgconfig: Remove the remaining bashisms
+	  in the Makefile/mkpkgconfig
+
+2009-09-24 08:33 +0000 [r220027]  Michiel van Baak <michiel at vanbaak.info>
+
+	* build_tools/mkpkgconfig: mkpkgconfig does not need bash so make
+	  it use /bin/sh This fixes building on all systems that don't have
+	  bash at /bin/bash Reported by _ys on #asterisk-dev Tested by _ys
+	  on #asterisk-dev
+
+2009-09-22 21:37 +0000 [r219816]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: When IMAP variables were changed during a
+	  reload, Voicemail did not use the new values. This change
+	  introduces a configuration version variable, which ensures that
+	  connections with the old values are not reused but are allowed to
+	  expire normally. (closes issue #15934) Reported by:
+	  viniciusfontes Patches: 20090922__issue15934.diff.txt uploaded by
+	  tilghman (license 14) Tested by: viniciusfontes
+
+2009-09-21 16:55 +0000 [r219720]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: Reverting merge 219520. This change was not
+	  necessary.
+
+2009-09-20 17:52 +0000 [r219653]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/file.c: Really stop the stream, when ast_closestream() is
+	  called. (closes issue #15129) Reported by: bmh Patches:
+	  20090918__issue15129.diff.txt uploaded by tilghman (license 14)
+	  Review: https://reviewboard.asterisk.org/r/372/
+
+2009-09-19 02:51 +0000 [r219586]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Make sure the iax_pvt exists before
+	  dereferencing it. This fixes the latest crash posted on issue
+	  15609. (issue #15609)
+
+2009-09-18 23:19 +0000 [r219450-219519]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: iax2 frame double free The iax frame's
+	  retrans sched id was written over right before iax2_frame_free
+	  was called. In iax2_frame_free that retrans id is used to delete
+	  the sched item. By writing over the retrans field before the
+	  sched item could be deleted, it was possible for a retransmit to
+	  occur on a freed frame.
+
+	* channels/chan_sip.c: via-header branches not updated correctly on
+	  INVITE INVITE requests must always contain a new unique branch
+	  id. When a new branch id is created for an INVITE, the dialog's
+	  invite_branch variable must be updated so CANCEL requests use the
+	  correct branch id. (closes issue #15262) Reported by: maniax
+	  Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety
+	  (license 608) invite_new_branch_trunk.diff uploaded by dvossel
+	  (license 671) Tested by: maniax, dvossel
+
+2009-09-17 22:20 +0000 [r219320]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Send a 100 Trying response when we detect a
+	  spiral. This was problematic during spiral tests at SIPit...
+	  along with some other things as well.
+
+2009-09-17 21:29 +0000 [r219303]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: INVITE w/Replaces deadlock fix This patch
+	  cleans up the locking logic in chan_sip.c's
+	  handle_invite_replaces() function as well as making use of

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