[asterisk-commits] bebuild: tag 1.6.2.23 r359764 - in /tags/1.6.2.23: .lastclean .version ChangeLog

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Mar 15 15:00:42 CDT 2012


Author: bebuild
Date: Thu Mar 15 15:00:38 2012
New Revision: 359764

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=359764
Log:
Importing files for 1.6.2.23 release.

Added:
    tags/1.6.2.23/.lastclean   (with props)
    tags/1.6.2.23/.version   (with props)
    tags/1.6.2.23/ChangeLog   (with props)

Added: tags/1.6.2.23/.lastclean
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.23/.lastclean?view=auto&rev=359764
==============================================================================
--- tags/1.6.2.23/.lastclean (added)
+++ tags/1.6.2.23/.lastclean Thu Mar 15 15:00:38 2012
@@ -1,0 +1,1 @@
+36

Propchange: tags/1.6.2.23/.lastclean
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.6.2.23/.lastclean
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.6.2.23/.lastclean
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.6.2.23/.version
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.23/.version?view=auto&rev=359764
==============================================================================
--- tags/1.6.2.23/.version (added)
+++ tags/1.6.2.23/.version Thu Mar 15 15:00:38 2012
@@ -1,0 +1,1 @@
+1.6.2.23

Propchange: tags/1.6.2.23/.version
------------------------------------------------------------------------------
    svn:eol-style = native

Propchange: tags/1.6.2.23/.version
------------------------------------------------------------------------------
    svn:keywords = none

Propchange: tags/1.6.2.23/.version
------------------------------------------------------------------------------
    svn:mime-type = text/plain

Added: tags/1.6.2.23/ChangeLog
URL: http://svnview.digium.com/svn/asterisk/tags/1.6.2.23/ChangeLog?view=auto&rev=359764
==============================================================================
--- tags/1.6.2.23/ChangeLog (added)
+++ tags/1.6.2.23/ChangeLog Thu Mar 15 15:00:38 2012
@@ -1,0 +1,31207 @@
+2012-03-15  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.6.2.23 Released.
+
+	* AST-2012-002
+
+2012-03-15 18:32 +0000 [r359645]  Matthew Jordan <mjordan at digium.com>
+
+	* apps/app_milliwatt.c, /: Fix remotely exploitable stack overrun
+	  in Milliwatt Milliwatt is vulnerable to a remotely exploitable
+	  stack overrun when using the 'o' option. This occurs due to the
+	  milliwatt_generate function not accounting for
+	  AST_FRIENDLY_OFFSET when calculating the maximum number of
+	  samples it can put in the output buffer. This patch resolves this
+	  issue by taking into account AST_FRIENDLY_OFFSET when determining
+	  the maximum number of samples allowed. Note that at no point is
+	  remote code execution possible. The data that is written into the
+	  buffer is the pre-defined Milliwatt data, and not custom data.
+	  (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
+	  by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
+	  Russell Bryant (license 6283) Note that this patch was written by
+	  Russell, even though Matt uploaded it
+
+2011-12-19  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.6.2.22 Released
+
+2011-12-18 18:25 +0000 [r348515]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/sip.conf.sample: Correct two flaws in sip.conf.sample
+	  related to AST-2011-013. 
+	  
+	  * The sample file listed *two* values
+	  for the 'nat' option as being the default. Only 'yes' is the
+	  default. 
+	  
+	  * The warning about having differing 'nat' settings
+	  confusingly referred to both peers and users.
+
+2011-12-08  Asterisk Development Team <asteriskteam at digium.com>
+
+	* Asterisk 1.6.2.21 Released.
+
+	* AST-2011-013, AST-2011-014
+
+2011-12-08 21:03 +0000 [r347659]  Leif Madsen <lmadsen at digium.com>
+
+	* /: Update svn:externals to use menuselect from 1.6.2.20 and not
+	  later. This change is required because when making security
+	  releases, if you pull from menuselect/trunk you'll get changes
+	  meant for later versions of Asterisk.
+
+2011-12-08 16:17 +0000 [r347530]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't crash on INFO automon request with no
+	  channel AST-2011-014. When automon was enabled in features.conf,
+	  it was possible to crash Asterisk by sending an INFO request if
+	  no channel had been created yet. (closes issue ASTERISK-18805)
+
+2011-11-21 20:33 +0000 [r345800-345827]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't set the nat default twice. Cleaning up
+	  a small merge issue ASTERISK-18862
+
+	* configs/sip.conf.sample, CHANGES, /, channels/chan_sip.c: Default
+	  to nat=yes; warn when nat in general and peer differ It is
+	  possible to enumerate SIP usernames when the general and
+	  user/peer nat settings differ in whether to respond to the port a
+	  request is sent from or the port listed for responses in the Via
+	  header. In 1.4 and 1.6.2, this would mean if one setting was
+	  nat=yes or nat=route and the other was either nat=no or
+	  nat=never. In 1.8 and 10, this would mean when one was
+	  nat=force_rport and the other was nat=no. In order to address
+	  this problem, it was decided to switch the default behavior to
+	  nat=yes/force_rport as it is the most commonly used option and to
+	  strongly discourage setting nat per-peer/user when at all
+	  possible. For more discussion of the issue, please see:
+	  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
+	  (closes issue ASTERISK-18862) Review:
+	  https://reviewboard.asterisk.org/r/1591/ ........ Merged
+	  revisions 345776 from
+	  http://svn.asterisk.org/svn/asterisk/branches/1.4
+
+2011-08-05  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.20 Released.
+
+2011-08-01 21:19 +0000 [r330490-330505]  Jonathan Rose <jrose at digium.com>
+
+	* main/features.c: fixes reference leak pointed out by rmudgett in
+	  https://reviewboard.asterisk.org/r/1337/
+
+	* main/features.c: Asterisk 18103 - Fix reload crash caused by
+	  destroying default parking lot Default parking lot was being
+	  destroyed in reload and was not being rebuilt properly. This
+	  patch keeps features.c reload from destroying the default parking
+	  lot in 1.6.2. Bug was caused by a hasty backport which didn't
+	  test reload enough times to catch the problem. (closes issue
+	  ASTERISK-18103) Reported by: 808blogger Review:
+	  https://reviewboard.asterisk.org/r/1337/
+
+2011-07-08 22:26 +0000 [r327255]  Jason Parker <jparker at digium.com>
+
+	* cdr, formats, codecs/gsm/src, funcs, bridges, codecs/lpc10,
+	  main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, res,
+	  pbx, res/ael, channels, main/stdtime, codecs, agi, utils,
+	  main/db1-ast/hash, apps, main/db1-ast/db, main/db1-ast/mpool: Add
+	  .o files to svn:ignore property, since it's only ignored if
+	  locally configured to do so.
+
+2011-06-28  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.6.2.19 Released.
+
+2011-06-28 20:06 +0000 [r325277]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 325275 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r325275 | twilson | 2011-06-28 15:03:19 -0500 (Tue, 28 Jun 2011)
+	  | 2 lines Don't leak SIP username information ........
+
+2011-06-23 18:21 +0000 [r324643]  Kinsey Moore <kmoore at digium.com>
+
+	* channels/chan_sip.c: Addresses AST-2011-008, memory corruption
+	  and remote crash in SIP driver. AST-2011-008
+
+2011-06-23 18:18 +0000 [r324634]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_iax2.c, /, main/features.c: Merged revisions 324627
+	  via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011)
+	  | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver
+	  Thanks to twilson for identifying the issue and providing the
+	  patches. AST-2011-010 ........
+
+2011-06-21 16:10 +0000 [r324306]  Kinsey Moore <kmoore at digium.com>
+
+	* apps/app_confbridge.c: ConfBridge does not handle hangup properly
+	  When playing back a prompt to a channel, confbridge neglects to
+	  check for hangup events causing lockup condititions for hangups
+	  that occur before actually joining the conference. This change
+	  ensures that the user is removed from the conference in the event
+	  of a premature hangup. Review:
+	  https://reviewboard.asterisk.org/r/1277/
+
+2011-06-15 18:13 +0000 [r323733]  Terry Wilson <twilson at digium.com>
+
+	* /, main/features.c: Merged revisions 323732 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011)
+	  | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a
+	  recent DTMF change. This patch makes sure that dynamic features
+	  are also checked when deciding whether or not to pass DTMF
+	  through or store it for interpreting. (closes issue
+	  ASTERISK-17914) Reported by: vrban ........
+
+2011-06-15 15:22 +0000 [r323579]  Sean Bright <sean at malleable.com>
+
+	* main/manager.c, /: Merged revisions 323559 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun
+	  2011) | 25 lines Resolve a segfault/bus error when we try to map
+	  memory that falls on a page boundary. The fix for ASTERISK-15359
+	  was incorrect in that it added 1 to the length of the mmap'd
+	  region. The problem with this is that reading/writing to that
+	  extra byte outside of the bounds of the underlying fd causes a
+	  bus error. The real issue is that we are working with both a FILE
+	  * and the raw fd underneath it and not synchronizing between
+	  them. The code that was removed in ASTERISK-15359 was correct,
+	  but we weren't flushing the FILE * before mapping the fd. Looking
+	  at the manager code in 1.4 reveals that the FILE * in 'struct
+	  mansession' is never used except to create a temporary file that
+	  we immediately fdopen. This means we just need to write a 0 byte
+	  to the fd and everything will just work. The other branches
+	  require a call to fflush() which, while not a guaranteed fix,
+	  should reduce the likelihood of a crash. This all makes sense in
+	  my head. (closes issue ASTERISK-16460) Reported by:
+	  Ravelomanantsoa Hoby (hoby) Patches:
+	  issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license
+	  #5060) ........
+
+2011-06-10 19:15 +0000 [r323039]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Unlock the sip channel during fax detection
+	  like chan_dahdi does to prevent a deadlock with
+	  ast_autoservice_stop. (closes issue ASTERISK-17798) tested by
+	  mnicholson
+
+2011-06-09 15:37 +0000 [r322668-322699]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: unlock pvt when we drop voice frames
+	  received in early media when in t.38 mode
+
+	* channels/chan_sip.c: fix for previous commit
+
+	* /, channels/chan_sip.c: Merged revisions 322646 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r322646 | mnicholson | 2011-06-09 10:10:30 -0500 (Thu, 09 Jun
+	  2011) | 5 lines don't drop any voice frames when checking for
+	  T.38 during early media (closes issue ASTERISK-17705) Review:
+	  https://reviewboard.asterisk.org/r/1186/ patch by oej reported by
+	  oej ........
+
+2011-05-27 08:24 +0000 [r321210]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/features.c: Fix *8 directed pickup locks system during
+	  pickupsound play out move playout from sip_pickup_thread to
+	  bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2
+	  threads trying to write audio to same channel. In addition fixes
+	  choppy audio beep in issue 19177. (issue #18654) (issue #19177)
+	  Reported by: Docent Patches: review1232-1.6.2.diff.txt uploaded
+	  by alecdavis (license 585) Tested by: alecdavis Review:
+	  https://reviewboard.asterisk.org/r/1232/
+
+2011-05-23 16:15 +0000 [r320506-320562]  David Vossel <dvossel at digium.com>
+
+	* main/tcptls.c: Adds missing part to the ast_tcptls_server_start
+	  fails second attempt to bind patch. (closes issue #19289)
+	  Reported by: wdoekes Patches:
+	  issue19289_delay_old_address_setting_tcptls_2.patch uploaded by
+	  wdoekes (license 717)
+
+	* apps/app_chanspy.c: Fixes chanspy enforced mode lacking a
+	  channel_unlock. (closes issue #19348) Reported by: wdoekes
+	  Patches: issue19348_chanspy_missing_channel_unlock.patch uploaded
+	  by wdoekes (license 717)
+
+2011-05-22 23:25 +0000 [r320444]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* res/res_odbc.c: Don't crash when the connection fails. (closes
+	  issue #19250) Reported by: seadweller Patches:
+	  20110514__issue19250.diff.txt uploaded by tilghman (license 14)
+	  Tested by: seadweller, sum
+
+2011-05-20 21:24 +0000 [r320271]  David Vossel <dvossel at digium.com>
+
+	* main/tcptls.c: Fixes issue with ast_tcptls_server_start failing
+	  on second attempt to bind. (closes issue #19289) Reported by:
+	  wdoekes Patches:
+	  issue19289_delay_old_address_setting_tcptls.patch uploaded by
+	  wdoekes (license 717)
+
+2011-05-20 20:44 +0000 [r320236]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, apps/app_meetme.c: Merged revisions 320235 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011)
+	  | 13 lines The meetme CLI command completion leaves conferences
+	  mutex locked. When issuing a meetme kick CLI command and an
+	  invalid (non-existent) conference number is specified, pressing
+	  Tab leaves the conferences mutex locked and, therefore, all
+	  conferences deadlock. Add missing unlock. (closes issue #19336)
+	  Reported by: zvision Patches: app_meetme.diff uploaded by zvision
+	  (license 798) ........
+
+2011-05-20 18:45 +0000 [r320179]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: This commit modifies the way polling is done
+	  on TLS sockets. Because of the buffering the TLS layer does,
+	  polling is unreliable. If poll is called while there is data
+	  waiting to be read in the TLS layer but not at the network layer,
+	  the messaging processing engine will not proceed until something
+	  else writes data to the socket, which may not occur. This change
+	  modifies the logic around TLS sockets to only poll after a failed
+	  read on a non-blocking socket. This way we know that there is no
+	  data waiting to be read from the buffering layer. (closes issue
+	  #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by
+	  mnicholson (license 96) Tested by: mnicholson
+
+2011-05-18 23:11 +0000 [r319528-319653]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 319652 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011)
+	  | 8 lines Make sure everyone gets an unhold when a transfer
+	  succeeds Some phones, like the Snom phones, send a hold to the
+	  transfer target after before sending the REFER. We need to make
+	  sure that we unhold the parties that are being connected after
+	  the masquerade. If Local channels with the /nm option are used
+	  when dialing the parties, hold music would still be playing on
+	  the transfer target, even after being connected with the
+	  transferee. ........
+
+	* apps/app_dial.c, /: Merged revisions 319527 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011)
+	  | 10 lines Fix app_dial ring groups Revert part of r315643. We
+	  need to remove the datastore here as well. The code in bridging
+	  code will catch anything that app_dial might miss. (closes issue
+	  #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff
+	  uploaded by elguero (license 37) ........
+
+2011-05-16 18:00 +0000 [r319202]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Unlink a peer from peers_by_ip when expiring
+	  a registration Review: https://reviewboard.asterisk.org/r/1218/
+
+2011-05-16 15:56 +0000 [r319144]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Fixes issue with peer ref-counting during
+	  handle_request_subscribe. (closes issue #19293) Reported by:
+	  irroot
+
+2011-05-16 15:51 +0000 [r319141]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Make sure tcptls_session exists before
+	  dereferencing it. (closes issue #19192) Reported by: stknob
+	  Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by
+	  Chainsaw (license 723) Tested by: vois, Chainsaw
+
+2011-05-13 01:14 +0000 [r318636-318735]  Richard Mudgett <rmudgett at digium.com>
+
+	* include/asterisk/features.h, /, channels/chan_sip.c,
+	  apps/app_directed_pickup.c, main/features.c: Merged revisions
+	  318734 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ................ r318734 | rmudgett | 2011-05-12 20:09:40 -0500
+	  (Thu, 12 May 2011) | 43 lines Merged revisions 318671 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 * The
+	  applicable fixes for v1.4 are the SIP deadlock and the in
+	  progress masquerade check for multiple parties trying to pickup
+	  the same call. issue18654_v1.4.patch uploaded by rmudgett
+	  (license 664) * Backported to v1.6.2. issue18654_v1.6.2.patch
+	  uploaded by rmudgett (license 664) ........ r318671 | alecdavis |
+	  2011-05-13 10:52:08 +1200 (Fri, 13 May 2011) | 30 lines Fix
+	  directed group pickup feature code *8 with pickupsounds enabled
+	  Since 1.6.2, the new pickupsound and pickupfailsound in
+	  features.conf cause many issues. 1).
+	  chan_sip:handle_request_invite() shouldn't be playing out the
+	  fail/success audio, as it has 'netlock' locked. 2). dialplan
+	  applications for directed_pickups shouldn't beep. 3). feature
+	  code for directed pickup should beep on success/failure if
+	  configured. Created a sip_pickup() thread to handle the pickup
+	  and playout the audio, spawned from handle_request_invite. Moved
+	  app_directed:pickup_do() to features:ast_do_pickup(). Functions
+	  below, all now use the new ast_do_pickup() app_directed_pickup.c:
+	  pickup_by_channel() pickup_by_exten() pickup_by_mark()
+	  pickup_by_part() features.c: ast_pickup_call() (closes issue
+	  #18654) Reported by: Docent Patches:
+	  ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license
+	  585) Tested by: lmadsen, francesco_r, amilcar, isis242,
+	  alecdavis, irroot, rymkus, loloski, rmudgett Review:
+	  https://reviewboard.asterisk.org/r/1185/ ........
+	  ................
+
+	* channels/chan_sip.c: Merged revision 222981 from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.8 Similar
+	  deadlock possible when running the Pickup application internally.
+	  ------------------------------------------------------------------------
+	  r222981 | dvossel | 2009-10-08 17:04:41 -0500 (Thu, 08 Oct 2009)
+	  | 13 lines Deadlock between ast_cel_report_event and
+	  ast_do_masquerade chan_sip calls pbx_exec on a pvt's owner
+	  channel while only the pvt lock is held. Since pbx_exec calls
+	  ast_cel_report_event which attempts to lock the channel, invalid
+	  locking order occurs. Channels should be locked before pvt's.
+	  (closes issue #15512) Reported by: lmsteffan Patches:
+	  ast_cel_deadlock_15512.diff uploaded by dvossel (license 671)
+
+2011-05-11 17:15 +0000 [r318548]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Clean up several chan_sip reference leaks
+	  Several situations in the code could lead to peers or sip_pvt
+	  references being leaked. This would cause RTP ports to never be
+	  destroyed (leading to exhaustion of all available RTP ports) and
+	  memory leaks. The original patch for this issue from rgagnon was
+	  the result of an obscene amount of testing and hard work, for
+	  which I am very grateful. I did some cleanup and added a few
+	  additional refcount fixes that I found. (closes issue #17255)
+	  Reported by: kvveltho Patches:
+	  tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff uploaded by
+	  rgagnon (license 1202) Tested by: rgagnon, twilson, wdoekes,
+	  loloski Review: https://reviewboard.asterisk.org/r/1101/ Review:
+	  https://reviewboard.asterisk.org/r/1207/
+
+2011-05-09 20:04 +0000 [r318331]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't offer video to directmedia callee
+	  unless caller offered it as well Make sure that when directmedia
+	  is enabled, that video is not offered to the callee even if it
+	  supports it. p->vrtp will not exist since the caller didn't offer
+	  video. (closes issue #19195) Reported by: one47 Patches:
+	  sip_cant_add_video_rtp uploaded by one47 (license 23)
+
+2011-05-09 16:51 +0000 [r318230]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: Fixes cases where sip_set_rtp_peer can
+	  return too early during media path reset. (closes issue #19225)
+	  Reported by: one47 Patches: sip_set_rtp_peer.patch uploaded by
+	  one47 (license 23)
+
+2011-05-06 19:34 +0000 [r317859]  Matthew Nicholson <mnicholson at digium.com>
+
+	* pbx/pbx_lua.c: pbx_lua autoservice fixes Don't start an
+	  autoservice in pbx_lua if pbx_lua already started one and don't
+	  stop one if we didn't start one. Also start and stop the
+	  autoservice when transferring control from and to the pbx.
+
+2011-05-06 18:03 +0000 [r317720]  Richard Mudgett <rmudgett at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 317719 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r317719 | rmudgett | 2011-05-06 12:59:05 -0500 (Fri, 06 May 2011)
+	  | 11 lines Regression after r297603 (Improve handling of REGISTER
+	  requests with multiple contact headers.) Uninitialized variable.
+	  (issue #18640) (closes issue #18785) Reported by: pnlarsson
+	  Patches: issue18785_enegaard.patch uploaded by enegaard (license
+	  1197) ........
+
+2011-05-06 15:18 +0000 [r317666]  Matthew Nicholson <mnicholson at digium.com>
+
+	* pbx/pbx_lua.c: Add a datastore fixup to fix a pbx_lua crash.
+	  (closes issue #19055) Reported by: jamhed Patches:
+	  lua_datastore_fixup1.diff uploaded by mnicholson (license 96)
+	  Tested by: mnicholson, jamhed
+
+2011-05-06 08:04 +0000 [r317575]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_queue.c, /: Merged revisions 317574 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011)
+	  | 6 lines Re-fix queue round-robin This part of the change for
+	  r315596 was incorrect. No bridge occurs when doing a roundrobin
+	  dial and no one answers, so this code shouldn't have been
+	  removed. ........
+
+2011-05-05 18:29 +0000 [r317255]  Russell Bryant <russell at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 317211 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011)
+	  | 15 lines chan_sip: fix broken realtime peer count, fix memory
+	  leak This patch addresses two bugs in chan_sip: 1) The count of
+	  realtime peers and users was off. The increment checked the value
+	  of the caching option, while the decrement did not. 2) Add a
+	  missing regfree() for a regex. (closes issue #19108) Reported by:
+	  vrban Patches: missing_regfree.patch uploaded by vrban (license
+	  756) sip_object_counter.patch uploaded by vrban (license 756)
+	  ........
+
+2011-05-05 17:59 +0000 [r317195]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Set SO_KEEPALIVE on SIP TCP sockets so that
+	  they eventually go away when a peer abruptly disappears. This
+	  mostly occurs after a successful registration. (closes issue
+	  #17544) Reported by: marcelloceschia Patches: (modified)
+	  tcptls.patch uploaded by st (license 907)
+
+2011-05-05 14:56 +0000 [r317103]  Leif Madsen <lmadsen at digium.com>
+
+	* contrib/scripts/safe_asterisk, /: Merged revisions 317102 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011)
+	  | 8 lines Disable console colourization inside safe_asterisk
+	  checks. (closes issue #19213) Reported by: lefoyer Patches:
+	  issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by
+	  wdoekes (license 717) Tested by: wdoekes, lefoyer ........
+
+2011-05-04 16:10 +0000 [r316708]  Sean Bright <sean at malleable.com>
+
+	* apps/app_voicemail.c, /: Merged revisions 316707 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed,
+	  04 May 2011) | 8 lines If sox fails when processing a voicemail,
+	  don't delete the original file. (closes issue #18111) Reported
+	  by: sysreq Patches: issue18111_trunk.patch uploaded by seanbright
+	  (license 71) Tested by: seanbright ........
+
+2011-05-04 14:23 +0000 [r316616-316644]  David Vossel <dvossel at digium.com>
+
+	* apps/app_chanspy.c: Fixes one-way-audio when chanspy activated
+	  with the 'o' option (closes issue #18382) Reported by: jkister
+	  Patches:
+	  0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt
+	  uploaded by malin (license ) Tested by: firstsip, Greenlightcrm,
+	  malin, wdoekes, boroda, dvossel
+
+	* channels/chan_sip.c: Fixes session-timers=refuse not being
+	  enforced for *caller* During handle_request_invite, the session
+	  timer mode was retrieved from a cached variable. This patch
+	  forces a peer lookup of the session timer mode in the case of an
+	  incoming invite. (closes issue #18804) Reported by: wdoekes
+	  Patches: issue18804_session_timer_refuse_caller.patch uploaded by
+	  wdoekes (license 717) issue_18804_v2.diff uploaded by dvossel
+	  (license 671)
+
+2011-05-04 02:23 +0000 [r316475]  Sean Bright <sean at malleable.com>
+
+	* apps/app_meetme.c: Honor the C option to MeetMe when L is passed.
+	  This fixes a case that r304773 and friends missed. (closes issue
+	  #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff
+	  uploaded by var (license 1227) Tested by: seanbright
+
+2011-05-03 21:29 +0000 [r316329]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_local.c, /: Merged revisions 316328 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03
+	  May 2011) | 10 lines Fixes chan_local crashs in local_fixup()
+	  Thanks OEJ for tracking down the issue and submitting the patch.
+	  (closes issue #19053) Reported by: oej Tested by: oej Review:
+	  https://reviewboard.asterisk.org/r/1158/ ........
+
+2011-05-02 19:04 +0000 [r316093]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* funcs/func_curl.c: More possible crashes based upon invalid
+	  inputs. (closes issue #18161) Reported by: wdoekes Patches:
+	  20110301__issue18161.diff.txt uploaded by tilghman (license 14)
+	  Tested by: wdoekes
+
+2011-04-27 19:03 +0000 [r315893]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 315891 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr
+	  2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2.
+	  This change optimizes the free_via() function and removes some
+	  redundant null checking. It also fixes compliance with RFC 3261
+	  section 18.2.2 by always using the port specified in the Via
+	  header for routing responses (even when maddr is not set). Also
+	  the htons() function is now used when setting the port.
+	  Additional documentation comments have been added in various
+	  places to make the logic in the code clearer. (closes issue
+	  #18951) Reported by: jmls Patches:
+	  issue18951_set_proper_port_from_via.patch uploaded by wdoekes
+	  (license 717) (modified) ........
+
+2011-04-26 22:52 +0000 [r315643-315672]  Terry Wilson <twilson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 315671 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011)
+	  | 11 lines Make sure unregistering a peer unlinks it from the
+	  peer container Instead of mostly copying the code from
+	  expire_register, just use the function that "does the right
+	  thing". (closes issue #16033) Reported by: kkm Patches:
+	  016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888)
+	  Tested by: kkm, tilghman, twilson ........
+
+	* apps/app_queue.c, apps/app_dial.c, /, main/features.c: Merged
+	  revisions 315596 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011)
+	  | 18 lines Allow transfer loops without allowing forwarding loops
+	  We try to avoid the situation where two phones may be forwarded
+	  to each other causing an infinite loop by storing each dialed
+	  interface in a channel datastore and checking the list before
+	  dialing out. This works, but currently breaks situations like A
+	  calls B, A transfers B to C, B transfers C to A, and A transfers
+	  C to B. Since human interaction is happening here and not an
+	  automated forwarding loop, it should be allowed. This patch
+	  removes the dialed_interfaces datastore when a call is bridged (a
+	  suggestion from the brilliant mmichelson). If a call is being
+	  bridged, it should be safe to assume that we aren't stuck in a
+	  loop. Since we are now handling this is the bridge code, the
+	  previous attempts at handling it in app_dial and app_queue are
+	  removed. Review: https://reviewboard.asterisk.org/r/1195/
+	  ........
+
+2011-04-26 19:22 +0000 [r315502]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* include/asterisk/select.h, /: Merged revisions 315501 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011)
+	  | 14 lines Fix the bounds-checking code. The code that set the
+	  bit within the select bitfield was correct, but the
+	  bounds-checking code was not. The change to that line uses the
+	  new _bitsize macro for clarity. Also, FD_ZERO macro did not
+	  zero-out anything but the first word of the bitfield, so this
+	  could have caused problems with modules using that macro with the
+	  expanded bitfield. (closes issue #18773) Reported by: jamicque
+	  Patches: 20110423__issue18773.diff.txt uploaded by tilghman
+	  (license 14) Tested by: chris-mac ........
+
+2011-04-26 02:17 +0000 [r315393]  Paul Belanger <pabelanger at digium.com>
+
+	* pbx/pbx_config.c: Add back CLI command 'dialplan save' (closes
+	  issue #19140) Reported by: lmadsen Patches:
+	  __20110419_dialplan_save.patch.txt uploaded by lmadsen (license
+	  10)
+
+2011-04-25 19:31 +0000 [r315212-315258]  Russell Bryant <russell at digium.com>
+
+	* /, formats/format_wav.c: Merged revisions 315257 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25
+	  Apr 2011) | 10 lines Be more flexible with unknown chunks in wav
+	  files. This patch makes format_wav ignore unknown chunks instead
+	  of erroring out on them. (closes issue #18306) Reported by:
+	  jhirsch Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch
+	  (license 1156) ........
+
+	* channels/chan_sip.c: Don't link non-cached realtime peers into
+	  the peers_by_ip container. (closes issue #18924) Reported by:
+	  wdoekes Patches:
+	  issue18924_uncached_realtime_peers_leak-1.6.2.17.patch uploaded
+	  by wdoekes (license 717)
+
+2011-04-25 07:11 +0000 [r315052]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_local.c, /: Merged revisions 315051 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25
+	  Apr 2011) | 11 lines chan_local:check_bridge() misplaced
+	  misplaced ast_mutex_unlock if !p->chan->_bridge->_softhangup path
+	  isn't followed, brigde remains locked. (closes issue #19176)
+	  Reported by: alecdavis Patches: bug19176.diff.txt uploaded by
+	  alecdavis (license 585) ........
+
+2011-04-22 20:49 +0000 [r314958]  Matthew Nicholson <mnicholson at digium.com>
+
+	* /, channels/chan_agent.c: Merged revisions 311203,314908 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar
+	  2011) | 4 lines Don't hold the pvt lock while streaming a file.
+	  ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48
+	  -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and
+	  the app threads from using the asterisk channel at the same time.
+	  ABE-2756 ........
+
+2011-04-22 14:35 +0000 [r314776-314823]  Russell Bryant <russell at digium.com>
+
+	* /: Recorded merge of revisions 314822 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r314822 | russell | 2011-04-22 09:34:23 -0500 (Fri, 22 Apr 2011)
+	  | 11 lines Initialize buffers in getvar and getvarfull.
+	  Initialize the buffers used to hold the result from GET VARIABLE
+	  or GET VARIABLE FULL. The bug report shows func_read returning
+	  garbage in the result. It assumed that the buffer passed in was
+	  initialized, like many other functions do. In the more common
+	  code path (through the dialplan), it is initialized, so just
+	  initialize it here too. (closes issue #19050) Reported by: johnz
+	  ........
+
+	* res/res_agi.c: Initialize buffers in getvar and getvarfull.
+	  Initialize the buffers used to hold the result from GET VARIABLE
+	  or GET VARIABLE FULL. The bug report shows func_read returning
+	  garbage in the result. It assumed that the buffer passed in was
+	  initialized, like many other functions do. In the more common
+	  code path (through the dialplan), it is initialized, so just
+	  initialize it here too. (closes issue #19050) Reported by: johnz
+
+	* main/features.c: Fix handling of some call parking config
+	  options. This patch adjusts the handling of some call parking
+	  config options to fix some issues that have already been
+	  addressed in 1.8 and trunk. (closes issue #19167) Reported by:
+	  bluecrow76 Patches:
+	  asterisk-1.6.2.17.2-fix-build-parkinglot-parked-AST_FEATURE_FLAGS.diff
+	  uploaded by bluecrow76 (license 270)
+
+2011-04-21 18:22 +0000 [r314620]  Matthew Nicholson <mnicholson at digium.com>
+
+	* configs/sip.conf.sample, configs/skinny.conf.sample,
+	  configs/http.conf.sample, main/manager.c, /, channels/chan_sip.c,
+	  channels/chan_skinny.c, main/http.c: Merged revisions 314607 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr
+	  2011) | 14 lines Added limits to the number of unauthenticated
+	  sessions TCP based protocols are allowed to have open
+	  simultaneously. Also added timeouts for unauthenticated sessions
+	  where it made sense to do so. Unrelated, the manager interface
+	  now properly checks if the user has the "system" privilege before
+	  executing shell commands via the Originate action. AST-2011-005
+	  AST-2011-006 (closes issue #18787) Reported by: kobaz (related to
+	  issue #18996) Reported by: tzafrir ........
+
+2011-04-21 00:17 +0000 [r314549]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Don't allocate more space than necessary for
+	  a sip_pkt This extra allocation is a hold-over from when
+	  pkt->data was a character array. Now that it is an allocated
+	  string, just allocate enough for the sip_pkt.
+
+2011-04-19 14:27 +0000 [r314202-314205]  Leif Madsen <lmadsen at digium.com>
+
+	* funcs/func_channel.c: Remove duplicate documentation from
+	  func_channel.c (closes issue #18970) Reported by: IgorG Patches:
+	  func_channel.c.doc.diff uploaded by IgorG (license 20)
+
+	* apps/app_dial.c: Update seconds to milliseconds in ast_verb
+	  output. (closes issue #19084) Reported by: smurfix Patches:
+	  app_dial.patch uploaded by smurfix (license 547) Tested by:
+	  lmadsen, smurfix
+
+2011-04-15 14:58 +0000 [r313859]  Jonathan Rose <jrose at digium.com>
+
+	* main/cli.c: Fix a Tab Completion bug that occurs due to multiple
+	  matches on a substring. Makes word_match function in cli.c repeat
+	  a search for a command string until a proper match is found or
+	  the string is searched to the last point. (closes issue #17494)
+	  Reported by: ffossard Review:
+	  https://reviewboard.asterisk.org/r/1180/
+
+2011-04-13 16:29 +0000 [r313579]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, /, res/res_agi.c: Merged revisions 313545 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011)
+	  | 41 lines Asterisk does not hangup a channel after endpoint
+	  hangs up. If the call that the dialplan started an AGI script for
+	  is hungup while the AGI script is in the middle of a command then
+	  the AGI script is not notified of the hangup. There are many AGI
+	  Exec commands that this can happen with. The reported
+	  applications have been: Background, Wait, Read, and Dial. Also
+	  the AGI Get Data command. * Don't wait on the Asterisk channel
+	  after it has hung up. The channel is likely to never need
+	  servicing again. * Restored the AGI script's ability to return
+	  the AGI_RESULT_HANGUP value in run_agi(). It previously only
+	  could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the
+	  DeadAGI and AGI applications were merged. (closes issue #17954)
+	  Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by
+	  rmudgett (license 664) issue17954_v1.6.2.patch uploaded by
+	  rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett
+	  (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue
+	  #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761
+	  (closes issue #18935) Reported by: nvitaly Tested by: astmiv,
+	  rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby
+	  Tested by: rmudgett JIRA SWP-2727 Review:
+	  https://reviewboard.asterisk.org/r/1165/ ........
+
+2011-04-12 18:44 +0000 [r313432-313435]  Jonathan Rose <jrose at digium.com>
+
+	* channels/chan_dahdi.c: fixing stupid mistake with putting code
+	  before variable declaration ........ Merged revisions 313433 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........
+	  r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) |
+	  14 lines reload Chan_dahdi memory leak caused by variables
+	  chan_dahdi reloading with variables set via setvar in
+	  chan_dahdi.conf would stay in the dahdi_pvt structs for
+	  individual channels (causing them to just continue adding the new
+	  ones to the list) and also there was a memory leak causes by the
+	  conf objects. This patch resolves both of these by using
+	  ast_variables_destroy during the loading process. (closes issue
+	  #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by
+	  jrose (license 1225) Tested by: tilghman, jrose Review:
+	  https://reviewboard.asterisk.org/r/1170/ ........ ........
+
+	* channels/chan_dahdi.c: white space change ........ reload
+	  Chan_dahdi memory leak caused by variables chan_dahdi reloading
+	  with variables set via setvar in chan_dahdi.conf would stay in
+	  the dahdi_pvt structs for individual channels (causing them to

[... 30472 lines stripped ...]



More information about the asterisk-commits mailing list