[asterisk-commits] lmadsen: tag 1.4.27-rc2 r222226 - /tags/1.4.27-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Oct 6 11:02:49 CDT 2009


Author: lmadsen
Date: Tue Oct  6 11:02:47 2009
New Revision: 222226

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=222226
Log:
Importing files for 1.4.27-rc2 release.

Added:
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    tags/1.4.27-rc2/.version   (with props)
    tags/1.4.27-rc2/ChangeLog   (with props)

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+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
+	  ast_do_masquerade() rather than forcing the masquerade on an
+	  ast_read(). The code had several redundant unlocks that would
+	  result in 'freed more times than we've locked!' errors. I cleaned
+	  these up as well as moving all the unlock logic to the end of the
+	  function. This patch should also resolve the issue people were
+	  having with the replacecall channel never being unlocked with one
+	  legged calls. (closes issue #15151) Reported by: irroot Patches:
+	  invite_w_replaces_1.4.diff uploaded by dvossel (license 671)
+	  Tested by: irroot, dvossel Review:
+	  https://reviewboard.asterisk.org/r/371/
+
+2009-09-17  Leif Madsen <lmadsen at digium.com>
+
+	* Release Asterisk 1.4.27-rc1
+
+2009-09-17 14:58 +0000 [r219136]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/channel.c, include/asterisk/cdr.h,
+	  include/asterisk/channel.h: Prevent a potential race condition
+	  and crash when hanging up a channel by removing the channel from
+	  the channel list before begining channel tear down. This fix may
+	  potentially cause problems with CDR backends that access the
+	  channel a CDR is associated with via the channel list. This fix
+	  makes the channel unavabile at the time when the CDR backend is
+	  invoked. This has been documented in include/asterisk/cdr.h.
+	  (closes issue #15316) Reported by: vmarrone Tested by: mnicholson
+	  Review: https://reviewboard.asterisk.org/r/362/
+
+2009-09-16 23:21 +0000 [r219023]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/config.c, configs/extensions.conf.sample: Properly deal with
+	  quotes in the arguments of '#exec' includes. (closes issue
+	  #15583) Reported by: pkempgen Patches:
+	  20090726__issue15583.diff.txt uploaded by tilghman (license 14)
+	  20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license
+	  169) Tested by: pkempgen
+
+2009-09-16 18:00 +0000 [r218867]  David Brooks <dbrooks at digium.com>
+
+	* main/pbx.c: Fixes CID pattern matching behavior to mirror that of
+	  extension pattern matching. Pattern matching for extensions uses
+	  a type of scoring system, giving values for specificity to each
+	  character in the pattern. Unfortunately, this is done character
+	  by character, in order. This does lead to some less specific
+	  patterns being first in line for matching, but it will usually
+	  get the job done. This patch merely brings CID matching to the
+	  same level as extension matching. This patch does not attempt to
+	  tackle the problem shared by extension matching. (closes issue
+	  #14708) Reported by: klaus3000
+
+2009-09-16 13:33 +0000 [r218798]  Russell Bryant <russell at digium.com>
+
+	* contrib/firmware/iax/iaxy.bin (removed), UPGRADE.txt: Remove the
+	  IAXy firmware from Asterisk. The firmware can now be found on
+	  downloads.digium.com, where the rest of our binary downloads
+	  live. This was the last part of our Asterisk tarballs that was
+	  considered non-free by Debian. :-) (closes issue #15838) Reported
+	  by: paravoid
+
+2009-09-15 22:27 +0000 [r218730]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: If the user enters the same password as
+	  before, don't signal an error when the change does nothing.
+	  (closes issue #15492) Reported by: cbbs70a Patches:
+	  20090713__issue15492.diff.txt uploaded by tilghman (license 14)
+
+2009-09-15 16:29 +0000 [r218623]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Fix small memory leak in handle_init_event
+	  by always destroying the pthread attr before returning.
+
+2009-09-15 16:03 +0000 [r218578]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Send request contact header field with
+	  response to registrer queries instead of the address of record.
+	  (closes issue #14438) Reported by: ravindrad Patches:
+	  regquerypatch uploaded by ravindrad (license 684) Tested by:
+	  ravindrad
+
+2009-09-15 16:01 +0000 [r218577]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_followme.c: Ensure FollowMe sets language in channels it
+	  creates. Also, not in the original bug report, but related fields
+	  are accountcode and musicclass, and the inheritance of
+	  datastores. (closes issue #15372) Reported by: Romik Patches:
+	  20090828__issue15372.diff.txt uploaded by tilghman (license 14)
+	  Tested by: cervajs
+
+2009-09-15 14:57 +0000 [r218497-218498]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: revert accidental commit
+
+	* channels/chan_sip.c, sounds/Makefile: Use proper hostname for
+	  downloading sound files.
+
+2009-09-14 21:47 +0000 [r218401]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Fix handling of DAHDI_EVENT_REMOVED event
+	  to prevent crash in do_monitor. After talking to rmudgett about
+	  some of his recent iflist locking changes, it was determined that
+	  the only place that would destroy a channel without being
+	  explicitly to do so was in handle_init_event. The loop to walk
+	  the interface list has been modified to wait to destroy the
+	  channel until the dahdi_pvt of the channel to be destroyed is no
+	  longer needed. (closes issue #15378) Reported by: samy
+
+2009-09-14 19:16 +0000 [r218331]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c, sounds/Makefile: Don't say "Please try
+	  again" if we don't give the user another chance to try again.
+	  (issue #15055, SWP-129) Reported by: jthurman
+
+2009-09-14 14:53 +0000 [r218223]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_directed_pickup.c: Ensure we don't pickup ourselves when
+	  doing pickup by exten. (closes issue #15100) Reported by:
+	  lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan
+	  (license 779)
+
+2009-09-10 23:52 +0000 [r217917-217989]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_queue.c: Don't ring another channel, if there's not
+	  enough time for a queue member to answer. (Fixes AST-228)
+
+	* contrib/scripts/iax-friends.sql, channels/chan_sip.c,
+	  channels/chan_iax2.c: Backport realtime fix to 1.4
+
+2009-09-10 21:06 +0000 [r217806]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c: IAX2 encryption regression The IAX2 Call
+	  Token security patch inadvertently broke the use of encryption
+	  due to the reorganization of code in the socket_process()
+	  function. When encryption is used, an incoming full frame must
+	  first be decrypted before the information elements can be parsed.
+	  The security release mistakenly moved IE parsing before
+	  decryption in order to process the new Call Token IE. To resolve
+	  this, decryption of full frames is once again done before looking
+	  into the frame. This involves searching for an existing callno,
+	  checking the pvt to see if encryption is turned on, and
+	  decrypting the packet before the internal fields of the full
+	  frame are accessed. associated with AST-2009-006 (closes issue
+	  #15834) Reported by: karesmakro Patches:
+	  iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671)
+	  Tested by: dvossel, karesmakro Review:
+	  https://reviewboard.asterisk.org/r/355/
+
+2009-09-10 19:52 +0000 [r217668-217735]  Olle Johansson <oej at edvina.net>
+
+	* utils/Makefile: Reinstate muted that was removed by mistake.
+	  muted doesn't compile any more on os/x, so I have to disable it
+	  in order to testcompile other code...
+
+	* utils/Makefile, channels/chan_sip.c: Remove harmful code that
+	  causes endless loops. Remove code that causes loops in
+	  registrations. We have agreed that the patch that this code was
+	  part of was bad. I am ripping out the code that causes the issue.
+	  putnopvut needs to check the rest of the patch, if it needs to be
+	  changed as well. This solves the issue reported in #15540, but
+	  needs more work before we close it (as described above).
+
+2009-09-08 20:01 +0000 [r217156]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_meetme.c: When MOH is playing on the channel,
+	  announcements sent through the conference are not heard. (closes
+	  issue #14588) Reported by: voipas Patches:
+	  20090716__issue14588__2.diff.txt uploaded by tilghman (license
+	  14) Tested by: lmadsen, twisted, tilghman
+
+2009-09-04 13:56 +0000 [r216432-216435]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/utils.c, include/asterisk/lock.h: make asterisk compile
+	  under devmode with DEBUG_THREADS enabled on OpenBSD
+
+	* channels/chan_sip.c: make chan_sip compile under devmode again
+
+2009-09-04 13:45 +0000 [r216430]  Olle Johansson <oej at edvina.net>
+
+	* apps/app_playback.c, main/pbx.c, channels/chan_sip.c,
+	  apps/app_disa.c, configs/sip.conf.sample: Make apps send PROGRESS
+	  control frame for early media and fix too early media issue in
+	  SIP The issue at hand is that some legacy (dying) PBX systems
+	  send empty media frames on PRI links *before* any call progress.
+	  The SIP channel receives these frames and by default signals 183
+	  Session progress and starts sending media. This will cause phones
+	  to play silence and ignore the later 180 ringing message. A bad
+	  user experience. The fix is twofold: - We discovered that
+	  asterisk apps that support early media ("noanswer") did not send
+	  any PROGRESS frame to indicate early media. Fixed. - We introduce
+	  a setting in chan_sip so that users can disable any relay of
+	  media frames before the outbound channel actually indicates any
+	  sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be
+	  disabled for backward compatibility. In later versions of
+	  Asterisk, this will be enabled. We don't assume that it will
+	  change your Asterisk phone experience - only for the better. We
+	  encourage third-party application developers to make sure that if
+	  they have applications that wants to send early media, add a
+	  PROGRESS control frame transmission to make sure that all channel
+	  drivers actually will start sending early media. This has not
+	  been the default in Asterisk previous to this patch, so if you
+	  got inspiration from our code, you need to update accordingly.
+	  Sorry for the trouble and thanks for your support. This code has
+	  been running for a few months in a large scale installation (over
+	  250 servers with PRI and/or BRI links to old PBX systems). That's
+	  no proof that this is an excellent patch, but, well, it's tested
+	  :-)
+
+2009-09-04 13:16 +0000 [r216369]  Michiel van Baak <michiel at vanbaak.info>
+
+	* main/astobj2.c: Make sure 'start' is always initialized. This is
+	  the same as rev 216222 in trunk but 1.4 is affected as well
+
+2009-09-04 10:48 +0000 [r216008-216263]  Russell Bryant <russell at digium.com>
+
+	* doc/IAX2-security.txt (added), /: Merged revisions 216262 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009)
+	  | 2 lines Add a plain text version of the IAX2 security document.
+	  ........
+
+	* /, UPGRADE.txt: Merged revisions 216080 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009)
+	  | 2 lines Add a note about IAX2 to UPGRADE.txt. ........
+
+	* /, doc/IAX2-security.pdf (added): Merged revisions 216005 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009)
+	  | 2 lines Add IAX2 security document related to AST-2009-006.
+	  ........
+
+2009-09-03 18:32 +0000 [r216000]  David Vossel <dvossel at digium.com>
+
+	* channels/iax2-parser.c, main/astobj2.c, configs/iax.conf.sample,
+	  include/asterisk/acl.h, channels/iax2-parser.h,
+	  include/asterisk/astobj2.h, channels/iax2.h, main/acl.c,
+	  channels/chan_iax2.c: Merge code associated with AST-2009-006
+	  (closes issue #12912) Reported by: rathaus Tested by: tilghman,
+	  russell, dvossel, dbrooks
+
+2009-09-02 21:41 +0000 [r215682]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Re-send non-100 provisional responses to
+	  prevent cancellation From section 13.3.1.1 of RFC 3261: If the
+	  UAS desires an extended period of time to answer the INVITE, it
+	  will need to ask for an "extension" in order to prevent proxies
+	  from canceling the transaction. A proxy has the option of
+	  canceling a transaction when there is a gap of 3 minutes between
+	  responses in a transaction. To prevent cancellation, the UAS MUST
+	  send a non-100 provisional response at every minute, to handle
+	  the possibility of lost provisional responses. (closes issue
+	  #11157) Reported by: rjain Tested by: twilson Review:
+	  https://reviewboard.asterisk.org/r/315/
+
+2009-09-01 23:04 +0000 [r215270]  Dwayne M. Hubbard <dwayne.hubbard at gmail.com>
+
+	* apps/app_softhangup.c: Use strrchr() so SoftHangup will correctly
+	  truncate multi-hyphen channel names In general channel names are
+	  in the form Foo/Bar-Z, but the channel name could have multiple
+	  hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the
+	  channel name at the last hyphen. (closes issue #15810) Reported
+	  by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by
+	  dhubbard (license 733)
+
+2009-08-31 16:16 +0000 [r214940]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_local.c: Also unlock the "other" channel, when
+	  returning, due to glare. (closes issue #15787) Reported by:
+	  tim_ringenbach Patches: chan_local.diff uploaded by tim
+	  ringenbach (license 540) Tested by: tim_ringenbach
+
+2009-08-28 20:13 +0000 [r214357-214701]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c: Modify comment to be a bit more accurate. We have
+	  kept this comment around long enough, that it's pretty clear that
+	  we're keeping the code, because changing the code would require a
+	  pretty fundamental architectural shift. We've also taken
+	  criticism in some quarters, because it was believed that it was
+	  referring to the code being nasty. No, the code isn't nasty, just
+	  the operation itself is rather odd. Fixed for eternity (probably
+	  not).
+
+	* autoconf/libcurl.m4 (added), configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Use autoconf to
+	  detect libcurl, as this enables cross-compilation checks,
+	  something we didn't allow before. (closes issue #15714) Reported
+	  by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded
+	  by tilghman (license 14) Tested by: pprindeville
+
+	* autoconf/ast_ext_lib.m4, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: One more build
+	  system change, to make the descriptions look better, if we have
+	  better information.
+
+	* autoconf/ast_ext_lib.m4, configure,
+	  include/asterisk/autoconfig.h.in: Make autoheader descriptions
+	  render correctly in our autoconfig.h file. (Figured out while
+	  working with issue #14906)
+
+2009-08-26 16:36 +0000 [r214194]  David Vossel <dvossel at digium.com>
+
+	* main/channel.c: ast_write() ignores ast_audiohook_write() results
+	  In ast_write(), if a channel has a list of audiohooks, those
+	  lists are written to and the resulting frame is what ast_write()
+	  should continue with. The problem was the returned audiohook
+	  frame was not being handled at all, and the original frame passed
+	  into it did not contain the mixed audio, so essentially audio was
+	  being lost. One result of this was chan_spy's whisper mode no
+	  longer worked. To complicate the issue, frames passed into
+	  ast_write may either be a single frame, or a list of frames. So,
+	  as the list of frames is processed in the audiohook_write, the
+	  returned frames had to be added to a new list. (closes issue
+	  #15660) Reported by: corruptor Tested by: dvossel
+
+2009-08-25 19:28 +0000 [r213899-214069]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/say.c: I should always compile before committing...
+
+	* main/say.c: Fix pronunciation of German dates. (closes issue
+	  #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded
+	  by Benjamin Kluck (license 803)
+
+	* main/pbx.c: Improve error message by informing user exactly which
+	  function is missing a parethesis. (closes issue #15242) Reported
+	  by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by
+	  dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by
+	  loloski (license 68)
+
+	* Makefile: Use the default runlevels for Debian derivatives,
+	  instead of making up our own. (closes issue #14730) Reported by:
+	  pkempgen
+
+2009-08-21 20:23 +0000 [r213631]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/chan_sip.c: Ensure that T.38 INVITEs generated by
+	  Asterisk properly result in T.38 being enabled. (closes issue
+	  #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by
+	  mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni
+
+2009-08-21 16:52 +0000 [r213559]  Tilghman Lesher <tlesher at digium.com>
+
+	* include/asterisk.h: Permit DEBUG_FD_LEAKS to be used with C++
+	  source files. (closes issue #15698) Reported by: slavon Patches:
+	  20090817__issue15698.diff.txt uploaded by tilghman (license 14)
+	  Tested by: slavon, tilghman
+
+2009-08-21 16:03 +0000 [r213493]  Jason Parker <jparker at digium.com>
+
+	* configs/queues.conf.sample: Clarify queues.conf comments to
+	  specify that variables should be set in the dialplan. (closes
+	  issue #15755) Reported by: trendboy
+
+2009-08-20 20:33 +0000 [r213339]  Matthew Nicholson <mnicholson at digium.com>
+
+	* res/res_features.c: Fix a crash by checking the proper pointer
+	  for validity before deferencing it. (closes issue #15751)
+	  Reported by: atis Patches: ast_bridge_call_peer_cdr.patch
+	  uploaded by atis (license 242)
+
+2009-08-20 19:53 +0000 [r213283]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.exports (added): Make all the symbols for the
+	  C-client callbacks global
+
+2009-08-19 21:18 +0000 [r213103]  David Vossel <dvossel at digium.com>
+
+	* apps/app_mixmonitor.c: Fixes memory leak caused by incorrectly
+	  freeing mixmonitor (closes issue #15699) Reported by: edantie
+	  Patches: mixmonitor.patch uploaded by edantie (license 862)
+
+2009-08-18 20:26 +0000 [r212913]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* doc/musiconhold-opsound.txt (added), CREDITS, /, UPGRADE.txt,
+	  sounds/sounds.xml, build_tools/prep_tarball,
+	  doc/musiconhold-fpm.txt (removed), doc/00README.1st,
+	  sounds/Makefile: Convert this branch to Opsound music-on-hold.
+	  For more details:
+	  http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
+
+2009-08-18 16:36 +0000 [r212763]  Sean Bright <sean at malleable.com>
+
+	* main/manager.c: Delay the creation of temporary files until we
+	  have a valid manager command to handle. Without this patch,
+	  asterisk creates a temporary file before determining if the
+	  specified command is valid. If invalid, we weren't properly
+	  cleaning up the file. (closes issue #15730) Reported by: zmehmood
+	  Patches: M15730.diff uploaded by junky (license 177) Tested by:
+	  zmehmood
+
+2009-08-18 16:00 +0000 [r212727]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c: Removed some deadwood and added some
+	  doxygen comments.
+
+2009-08-17 16:34 +0000 [r212498]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/misdn_config.c: Fix segfault when reloading chan_misdn.
+	  If more ports were specified than configured in misdn.conf a
+	  reload would crash asterisk. The problem was the unconfigured
+	  port was using data from the previously configured port. When the
+	  data for an unconfigured port was freed a crash would result from
+	  the double free. (closes issue #12113) Reported by: agupta
+	  Patches: bug12113.patch uploaded by jpeeler (license 325)
+
+2009-08-17 15:36 +0000 [r212430]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Fix uninitialized variable.
+
+2009-08-12 23:04 +0000 [r211953]  Matthew Nicholson <mnicholson at digium.com>
+
+	* apps/app_queue.c: This patch adds additional checking when
+	  generating queue log TRANSFER events. The additional checks
+	  prevent generation of false TRANSFER events in certain
+	  situations. (closes issue #14536) Reported by: aragon Patches:
+	  queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
+	  Tested by: aragon, mnicholson
+
+2009-08-12 18:46 +0000 [r211807]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Backport fix so that outbound CANCEL
+	  requests have same branch as challenged INVITEs. There already
+	  was code present to be sure that a CANCEL will contain the same
+	  branch-id as the INVITE it is cancelling. However, for INVITES
+	  which are challenged downstream, this mechanism did not work
+	  properly. Now this is taken care of. This is a backport of a fix
+	  already present in all 1.6.X branches and in trunk. It also fixes
+	  ABE-1907.
+
+2009-08-10 19:48 +0000 [r211528-211583]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/CODING-GUIDELINES: Conversion specifiers, not format
+	  specifiers
+
+	* main/indications.c, main/cli.c, pbx/pbx_loopback.c,
+	  channels/chan_dahdi.c, res/res_smdi.c, pbx/pbx_spool.c,
+	  channels/chan_skinny.c, pbx/pbx_ael.c, apps/app_dial.c,
+	  main/pbx.c, apps/app_privacy.c, codecs/codec_speex.c,
+	  funcs/func_math.c, channels/chan_agent.c, apps/app_morsecode.c,
+	  apps/app_disa.c, channels/iax2-provision.c, funcs/func_cut.c,
+	  pbx/dundi-parser.c, apps/app_talkdetect.c, channels/chan_misdn.c,
+	  apps/app_waitforring.c, funcs/func_channel.c, apps/app_macro.c,
+	  pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c,
+	  main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+	  doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+	  main/utils.c, apps/app_followme.c, utils/frame.c,
+	  channels/misdn_config.c, main/cdr.c, main/channel.c,
+	  channels/chan_phone.c, main/manager.c, apps/app_osplookup.c,
+	  apps/app_setcallerid.c, res/res_agi.c, apps/app_rpt.c,
+	  channels/chan_mgcp.c, apps/app_adsiprog.c, main/dnsmgr.c,
+	  channels/chan_sip.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+	  main/acl.c, apps/app_queue.c, channels/chan_oss.c,
+	  agi/eagi-sphinx-test.c, channels/chan_h323.c, pbx/pbx_dundi.c,
+	  apps/app_sms.c, apps/app_verbose.c, apps/app_dahdibarge.c,
+	  funcs/func_rand.c, apps/app_readfile.c, main/frame.c, /,
+	  res/res_features.c, apps/app_record.c, funcs/func_strings.c,
+	  apps/app_random.c, apps/app_alarmreceiver.c,
+	  channels/chan_iax2.c: AST-2009-005
+
+2009-08-09 15:41 +0000 [r211274]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/astfd.c: Small oops. Clear the flags which have been
+	  checked.
+
+2009-08-07 20:11 +0000 [r211112]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c: Resolve a deadlock involving app_chanspy and
+	  masquerades. (ABE-1936)
+
+2009-08-07 18:16 +0000 [r210913-211038]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_queue.c: QUEUE_MEMBER_LIST _really_ wants the interface
+	  name, not the membername. This is a partial revert of revision
+	  82590, which was an attempted cleanup, but in reality, it broke
+	  QUEUE_MEMBER_LIST, which has always been intended as a method by
+	  which component interfaces could be queried from the queue.
+	  Membername isn't useful here, because that field cannot be used
+	  to obtain further information about the member. See the
+	  documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+	  QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+	  member argument for further justification. (closes issue #15664)
+	  Reported by: rain Patches: app_queue-queue_member_list.diff
+	  uploaded by rain (license 327)
+
+	* main/channel.c: Because channel information can be accessed
+	  outside of the channel thread, we must lock the channel prior to
+	  modifying it. (closes issue #15397) Reported by: caspy Patches:
+	  20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+	  Tested by: caspy
+
+2009-08-05 19:18 +0000 [r210575]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Dialplan starts execution before the
+	  channel setup is complete. * Issue 15655: For the case where
+	  dialing is complete for an incoming call, dahdi_new() was asked
+	  to start the PBX and then the code set more channel variables. If
+	  the dialplan hungup before these channel variables got set,
+	  asterisk would likely crash. * Fixed potential for overlap
+	  incoming call to erroneously set channel variables as global
+	  dialplan variables if the ast_channel structure failed to get
+	  allocated. * Added missing set of CALLINGSUBADDR in the dialing
+	  is complete case. (closes issue #15655) Reported by: alecdavis
+
+2009-08-05 18:46 +0000 [r210563]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/imapstorage.txt: Update imapstorage.txt documentation.
+	  Updated the imapstorage.txt documentation to reflect that issues
+	  with c-client versions older than 2007 seem to cause crashing
+	  issues that are not seen with more recent versions. Documentation
+	  has been updated to reflect this. (closes issue #14496) Reported
+	  by: vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+	  uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+	  dbrooks
+
+2009-08-04 14:51 +0000 [r210237]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile: Eliminate spurious compiler warnings from system
+	  headers on *BSD platforms. Ensure that system headers located in
+	  /usr/local/include are actually treated as system headers by the
+	  compiler, and not as local headers which are subject to warnings
+	  from the -Wundef compiler option and others. (closes issue
+	  #15606) Reported by: mvanbaak
+
+2009-08-03 16:15 +0000 [r210067]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_dahdi.c: Fixes dialplan wildcard extension taking
+	  precedence over call pickup code. Prior to this patch, a wildcard
+	  extension in the dialplan (for example, _*.) would take
+	  precedence over picking up a call in the channel's pickup group.
+	  This patch simply moves the block of code handling pickup group
+	  matching to above the extension matching code. (closes issue
+	  #14735) Reported by: stevedavies Review:
+	  https://reviewboard.asterisk.org/r/319/
+
+2009-08-03 16:11 +0000 [r210064-210066]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_oss.c, apps/app_playback.c, main/asterisk.exports,
+	  configure, include/asterisk/autoconfig.h.in,
+	  include/asterisk/compat.h, main/strcompat.c, configure.ac,
+	  funcs/func_cut.c: Reverting index() fix, applying a different
+	  methodology, based upon developer discussions. (related to issue
+	  #15639)
+
+	* main/asterisk.exports, include/asterisk/compat.h: Helps if we
+	  export the index() function. (Related to issue #15639)
+
+	* configure, include/asterisk/autoconfig.h.in, main/strcompat.c,
+	  configure.ac: Apparently, some platforms don't have the index()
+	  function. (closes issue #15639) Reported by: nmav
+
+2009-08-01 11:27 +0000 [r209838-209879]  Russell Bryant <russell at digium.com>
+
+	* main/db1-ast/mpool/mpool.c: Resolve a valgrind warning about a
+	  read from uninitialized memory. (issue #15396) Reported by:
+	  aragon
+

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