[asterisk-commits] lmadsen: tag 1.4.42-rc2 r324112 - /tags/1.4.42-rc2/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Jun 17 08:51:45 CDT 2011


Author: lmadsen
Date: Fri Jun 17 08:51:41 2011
New Revision: 324112

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=324112
Log:
Importing files for 1.4.42-rc2 release.

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    tags/1.4.42-rc2/ChangeLog   (with props)

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--- tags/1.4.42-rc2/ChangeLog (added)
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+2011-06-17  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.42-rc2 released.
+
+2011-06-15 18:06 +0000 [r323732]  Terry Wilson <twilson at digium.com>
+
+	* res/res_features.c: 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 15:15 +0000 [r323559]  Sean Bright <sean at malleable.com>
+
+	* main/manager.c: 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)
+
+2011-06-09 15:36 +0000 [r322646-322698]  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: whitespace
+
+	* channels/chan_sip.c: 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
+
+2011-05-21 05:09 +0000 [r320393]  Paul Belanger <pabelanger at digium.com>
+
+	* cdr/cdr_pgsql.c: Solaris compatibility fixes
+
+2011-05-20 20:38 +0000 [r320235]  Richard Mudgett <rmudgett at digium.com>
+
+	* apps/app_meetme.c: 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 16:38 +0000 [r320055]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_sip.c: chan_sip: Destroy variables on a sip_pvt
+	  before copying vars from the sip_peer. (closes issue #19202)
+	  Reported by: wdoekes Patches:
+	  issue19202_destroy_challenged_invite_chanvars.patch uploaded by
+	  wdoekes (license 717)
+
+2011-05-18 23:04 +0000 [r319527-319652]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: 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: 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  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.42-rc1
+
+	This will be the final set of release candidates for the Asterisk
+	1.4 branch.
+
+2011-05-13 01:09 +0000 [r318734]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c, res/res_features.c,
+	  apps/app_directed_pickup.c: 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/ ........
+
+2011-05-06 17:59 +0000 [r317719]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_sip.c: 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 07:55 +0000 [r317574]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_queue.c: 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 18:20 +0000 [r317211]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: 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 14:54 +0000 [r317102]  Leif Madsen <lmadsen at digium.com>
+
+	* contrib/scripts/safe_asterisk: 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 16:08 +0000 [r316707]  Sean Bright <sean at malleable.com>
+
+	* apps/app_voicemail.c: 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
+
+2011-05-03 21:27 +0000 [r316328]  David Vossel <dvossel at digium.com>
+
+	* channels/chan_local.c: 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 18:25 +0000 [r316089]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* configure, configure.ac: Breakage from slightly before the
+	  outage; would have fixed sooner but for the outage.
+
+2011-04-27 21:20 +0000 [r316006]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac:
+	  Backport the use of curl from 1.6.2 to make the 1.4 target work
+	  on Bamboo.
+
+2011-04-27 20:54 +0000 [r315989]  Sean Bright <sean at malleable.com>
+
+	* channels/chan_sip.c: Partial revert of r315671 which removed a
+	  logging statement and not a manager event. Reported by ibercom in
+	  #asterisk-bugs. (issue #16033)
+
+2011-04-27 18:57 +0000 [r315891]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: 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 22:47 +0000 [r315596-315671]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: 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_dial.c, res/res_features.c, apps/app_queue.c: 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 19:18 +0000 [r315501]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* include/asterisk/select.h: 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
+
+2011-04-25 19:28 +0000 [r315257]  Russell Bryant <russell at digium.com>
+
+	* formats/format_wav.c: 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)
+
+2011-04-25 16:14 +0000 [r315147]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/manager.c: Reverted part of r314607, as it can introduce a
+	  regression. Specifically, the security check for the "system"
+	  privilege was removed. If a user had the "call" privilege but not
+	  the "system" privilege, they would loose the ability to execute
+	  the system app and dialplan functions that run commands in a
+	  shell. This branch never used the "system" privilege for that
+	  purpose and did not need to be patched. AST-2011-006 (related to
+	  issue 0018787) Reported by: kobaz
+
+2011-04-25 07:06 +0000 [r315051]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* channels/chan_local.c: 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 20:01 +0000 [r314908]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_agent.c: Prevent the login thread and the app
+	  threads from using the asterisk channel at the same time.
+	  ABE-2756
+
+2011-04-22 14:34 +0000 [r314822]  Russell Bryant <russell at digium.com>
+
+	* 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
+
+2011-04-21 18:19 +0000 [r314607]  Matthew Nicholson <mnicholson at digium.com>
+
+	* configs/http.conf.sample, main/manager.c, channels/chan_skinny.c,
+	  main/http.c, configs/skinny.conf.sample: 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
+
+2011-04-19 18:37 +0000 [r314300]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* apps/app_voicemail.c: app_voicemail: Fix ODBC Storage compile
+	  regression caused by me, from mantis bug #19032 / commit r312070
+	  (closes issue #19142) Reported by: vrban Patches:
+	  app_voicemail_fix_for_312070.patch uploaded by vrban (license
+	  756) Tested by: vrban, alecdavis
+
+2011-04-13 16:21 +0000 [r313545]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, res/res_agi.c: 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/
+
+2011-04-11 19:30 +0000 [r313277]  Leif Madsen <lmadsen at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac: Fix
+	  detection of OpenSSL 1.0 (closes issue #19093) Reported by:
+	  tzafrir Patches: detect_openssl_10.diff uploaded by tzafrir
+	  (license 46)
+
+2011-04-11 15:27 +0000 [r313188]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Stuck channel using FEATD_MF if caller
+	  hangs up at the right time. The cause was actually a caller
+	  hanging up just at the end of the Feature Group D DTMF tones that
+	  setup the call. The reason for this is a "guard timer" that's
+	  implemented using ast_safe_sleep(100). If the caller happens to
+	  hang up AFTER the final tone of the DTMF string but BEFORE the
+	  end of that ast_safe_sleep(), then ast_safe_sleep() will return
+	  non-zero. This causes the code to bounce to the end of
+	  ss_thread(), but it does NOT tear down the call properly. This
+	  should be a rare occurrence because the caller has to hang up at
+	  EXACTLY the right time. Nonetheless, it was happening quite
+	  regularly on the reporter's system. It's not easily reproducible,
+	  unless you purposely increase the guard-time to 2000 or more.
+	  Once you do that, you can reproduce it every time by watching the
+	  DTMF debug and hanging up just as it ends. Simply add an
+	  ast_hangup() before goto quit. (closes issue #15671) Reported by:
+	  jcromes Patches: issue15671.patch uploaded by pabelanger (license
+	  224) Tested by: jcromes
+
+2011-04-05 14:10 +0000 [r312761]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/manager.c, configs/manager.conf.sample: Limit the number of
+	  unauthenticated manager sessions and also limit the time they
+	  have to authenticate. AST-2011-005 (closes issue #18996) Reported
+	  by: tzafrir Tested by: mnicholson
+
+2011-04-04 15:49 +0000 [r312573]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Issues with ISDN
+	  calls changing B channels during call negotiations. The handling
+	  of the PROCEEDING message was not using the correct call
+	  structure if the B channel was changed. (The same for PROGRESS.)
+	  The call was also not hungup if the new B channel is not
+	  provisioned or is busy. * Made all call connection messages
+	  (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, ALERTING, CONNECT,
+	  CONNECT_ACKNOWLEDGE) ensure that they are using the correct
+	  structure and B channel. If there is any problem with the
+	  operations then the call is now hungup with an appropriate cause
+	  code. * Made miscellaneous messages (INFORMATION, FACILITY,
+	  NOTIFY) find the correct structure by looking for the call and
+	  not using the channel ID. NOTIFY is an exception with versions of
+	  libpri before v1.4.11 because a call pointer is not available for
+	  Asterisk to use. * Made all hangup messages (DISCONNECT, RELEASE,
+	  RELEASE_COMPLETE) find the correct structure by looking for the
+	  call and not using the channel ID. (closes issue #18313) Reported
+	  by: destiny6628 Tested by: rmudgett JIRA SWP-2620 (closes issue
+	  #18231) Reported by: destiny6628 Tested by: rmudgett JIRA
+	  SWP-2924 (closes issue #18488) Reported by: jpokorny JIRA
+	  SWP-2929 JIRA AST-437 (The issues fixed here are most likely
+	  causing this JIRA issue.) JIRA DAHDI-406 JIRA LIBPRI-33 (Stuck
+	  resetting flag likely fixed)
+
+2011-04-01 11:02 +0000 [r312290]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* apps/app_voicemail.c: app_voicemail: leave_vociemail doesn't use
+	  last_message_index to store next message trivial change to bring
+	  inline with 1.6.2 1.8 and trunk. The symptom was if msg0000 was
+	  missing, and the last was msg0004, the next msgnum would be
+	  msg0000 when it should have been msg0005 (issue #18998) Reported
+	  by: tootai Patches: bug18998.diff2.txt uploaded by alecdavis
+	  (license 585) Tested by: alecdavis
+
+2011-04-01 10:36 +0000 [r312285]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* main/asterisk.c, include/asterisk/select.h: Found some leaking
+	  file descriptors while looking at ast_FD_SETSIZE dead code.
+	  (issue #18969) Reported by: oej Patches:
+	  20110315__issue18969__14.diff.txt uploaded by tilghman (license
+	  14)
+
+2011-04-01 08:29 +0000 [r312070-312174]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* apps/app_voicemail.c: voicemail: get real last_message_index and
+	  count_messages, ODBC resequence change last_message_index to read
+	  the max msgnum stored in the database change count_messages to
+	  actually count the number of messages. last_message_index change:
+	  This fixed overwriting of the last message if msgnum=0 was
+	  missing. Previously every incoming message would overwrite
+	  msgnum=1. count_messages change: allows us to detect when
+	  requencing is required in opneA_mailbox. resequence enabled for
+	  ODBC storage: Assists with fixing up corrupt databases with gaps,
+	  but only when a user actively opens there mailboxes. (closes
+	  issue #18692,#18582,#19032) Reported by: elguero Patches: based
+	  on odbc_resequence_mailbox2.1.diff uploaded by elguero (license
+	  37) Tested by: elguero, nivek, alecdavis Review:
+	  https://reviewboard.asterisk.org/r/1153/
+
+	* apps/app_voicemail.c: app_voicemail:close_mailbox imap_storage
+	  doesn't use last_msg_index
+
+	* apps/app_voicemail.c: app_voicemail: close_mailbox needs to
+	  respect additional messages while mailbox is open. close_mailbox
+	  leave gaps in message sequence if messages are deleted and new
+	  messages arrive during this time, this is because the shuffle
+	  down to slot 0, only shuffles the number of pre-existing messages
+	  when mailbox is opened, ignoring new arrivals. Fix: in
+	  close_mailbox re-evaluate number of messages before the shuffle,
+	  this then includes new arrivals. Happens on filebased or ODBC
+	  storage. (issues #19032,#18582,#18692,#18998) Reported by:
+	  alecdavis,tootai,afosorio Review:
+	  https://reviewboard.asterisk.org/r/1153/
+
+2011-03-17 19:14 +0000 [r311199-311203]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_agent.c: Don't hold the pvt lock while streaming a
+	  file. ABE-2756
+
+	* main/manager.c: Don't dec the usecount of an eventqent then use
+	  it. ABE-2756
+
+	* channels/chan_sip.c: Remove the provisional keepalive scheduler
+	  entry's reference to the pvt when we remove the scheduler entry.
+	  ABE-2756
+
+2011-03-17 10:43 +0000 [r311048]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* configs/indications.conf.sample: Remove extra quote in
+	  indications.conf Picking low hanging fruit. (closes issue #18971)
+	  Reported by: IgorG Patches: based on indications.conf.sample.diff
+	  uploaded by IgorG (license 20) Tested by: IgorG
+
+2011-03-16 16:58 +0000 [r310888]  Terry Wilson <twilson at digium.com>
+
+	* res/res_features.c: Don't delay DTMF in core bridge while
+	  listening for DTMF features This patch is mostly the work of Olle
+	  Johansson. I did some cleanup and added the silence generating
+	  code if transmit_silence is set. When a channel listens for DTMF
+	  in the core bridge, the outbound DTMF is not sent until we have
+	  received DTMF_END. For a long DTMF, this is a disaster. We send 4
+	  seconds of DTMF to Asterisk, which sends no audio for those 4
+	  seconds. Some products see this delay and the time skew on RTP
+	  packets that results and start ignoring the audio that is sent
+	  afterward. With this change, the DTMF_BEGIN frame is inspected
+	  and checked. If it matches a feature code, we wait for DTMF_END
+	  and activate the feature as before. If transmit_silence=yes in
+	  asterisk.conf, silence is sent if we paritally match a
+	  multi-digit feature. If it doesn't match a feature, the frame is
+	  forwarded along with the DTMF_END without delay. By doing it this
+	  way, DTMF is not delayed. (closes issue #15642) Reported by:
+	  jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded
+	  by twilson (license 396) Tested by: globalnetinc, jde (closes
+	  issue #16625) Reported by: sharvanek Review:
+	  https://reviewboard.asterisk.org/r/1092/ Review:
+	  https://reviewboard.asterisk.org/r/1125/
+
+2011-03-15 00:26 +0000 [r310779]  Alec L Davis <sivad.a at paradise.net.nz>
+
+	* main/utils.c: core show locks: display ThreadID in hexadecimal
+	  Allow easier cross referencing of thread ID's with GDB backtraces
+	  (closes issue #18968) Reported by: alecdavis Patches:
+	  bug18968.diff.txt uploaded by alecdavis (license 585)
+
+2011-03-14 16:38 +0000 [r310633]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/callerid.c: "Caller*ID failed checksum" on Wildcard TDM2400P
+	  and TDM410 The last character in the caller id message is getting
+	  a framing error. The checksum is the last character in the
+	  message. A framing error in the checksum could be because: 1) The
+	  sender did not send a full stop bit. 2) The sender cut off the
+	  FSK carrier too soon. 3) The sender opted to send zero of the
+	  specified zero to 10 trailing mark bits and round-off errors in
+	  the code resulted in the code not being where it thought it was
+	  in the demodulated bit stream. Bit 8 of 'b' is set when parity
+	  error. Bit 9 of 'b' is set when framing error. Made ignore the
+	  framing and parity error bits if the errored character is the
+	  checksum. We can tolerate a framing/parity error there. The
+	  checksum character validates the message. (closes issue #18474)
+	  Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek
+	  (license 636) (with modifications) Tested by: nivek
+
+2011-03-12 20:22 +0000 [r310435]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* pbx/pbx_ael.c: Add AELSub, which provides a stable entry point
+	  into AEL subroutines. This commit needs some explanation, given
+	  that we're adding a new application into an existing release
+	  branch. This is generally a violation of our release policy,
+	  except in very limited circumstances, and I believe this is one
+	  of those circumstances. The problem that this solves is one of
+	  the sanity of using multiple dialplan languages to define a
+	  dialplan. In the case of the reporter, he or she is using AEL is
+	  define subroutines, while using Realtime extensions to invoke
+	  those subroutines. While you can do this, it's based upon the
+	  reality of AEL using actual dialplan extensions; however, there
+	  is no guarantee that the details of _how_ AEL is compiled into
+	  extensions will remain stable. In fact, at the time of this
+	  commit, it has already changed twice, once in a fundamental way.
+	  Now normally, a new application would only be added to trunk.
+	  However, this application is explicitly to create a stable
+	  user-level API between versions, and adding it to trunk only will
+	  not solve the user's problem of switching between 1.6.2 and 1.8,
+	  nor will it help anybody switching from 1.8 to 1.10. Therefore,
+	  it needs to go into existing release branches. For the sake of
+	  consistency, and also because one of the changes was between 1.4
+	  and 1.6.x, I am also electing to commit this to 1.4. (closes
+	  issue #18910) Reported by: alexandrekeller Patches:
+	  20110304__issue18919__1.6.2.diff.txt uploaded by tilghman
+	  (license 14) 20110304__issue18919__1.4.diff.txt uploaded by
+	  tilghman (license 14) Tested by: alexandrekeller
+
+2011-03-10 05:38 +0000 [r310140]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* apps/app_voicemail.c, res/res_config_odbc.c: Initialize column
+	  size to 0 to deal with a potential UnixODBC bug on 64-bit
+	  systems. (closes issue #18295) Reported by: pruiz
+
+2011-03-08 02:42 +0000 [r309947]  Terry Wilson <twilson at digium.com>
+
+	* apps/app_externalivr.c: Don't try to free statically allocated
+	  memory. Note: compiling after ./configure --enable-dev-mode will
+	  keep these kinds of mistakes from being committed.
+
+2011-03-07 22:02 +0000 [r309856]  Jonathan Rose <jrose at digium.com>
+
+	* apps/app_mixmonitor.c: Bug fix for MixMonitor involving filenames
+	  with '.' not in the extension Closes issue #18391) Reported by:
+	  pabelanger Patches: bugfix.patch uploaded by jrose (license 1225)
+	  Tested by: jrose
+
+2011-03-04 00:34 +0000 [r309355]  David Ruggles <thedavidfactor at gmail.com>
+
+	* apps/app_externalivr.c: fix small memory leak fix small memory
+	  leak caused by a string allocation that wasn't freed (closes
+	  issue #18907) Reported by: andy11 Patches:
+	  asterisk_trunk-app_externalivr-leak.patch uploaded by andy11
+	  (license 1224)
+
+2011-02-24 17:42 +0000 [r308813]  Terry Wilson <twilson at digium.com>
+
+	* main/manager.c: Don't broadcast FullyBooted to every AMI
+	  connection The FullyBooted event should not be sent to every AMI
+	  connection every time someone connects via AMI. It should only be
+	  sent to the user who just connected. (closes issue #18168)
+	  Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre
+	  (license 1142) Tested by: FeyFre, twilson
+
+2011-02-24 14:54 +0000 [r308721]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/udptl.c: silence gcc 4.2 compiler warning
+
+2011-04-25  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.41 Released.
+
+	* AST-2011-005, AST-2011-006
+
+	* Reverted part of r314607, as it can introduce a regression.
+	  Specifically, the security check for the "system" privilege was
+	  removed. If a user had the "call" privilege but not the "system" privilege,
+	  they would lose the ability to execute the system app and dialplan functions
+	  that run commands in a shell. This branch never used the "system" privilege
+	  for that purpose and did not need to be patched.
+	  (AST-2011-006)
+
+2011-02-23  Leif Madsen <lmadsen at digium.com>
+
+	* Asterisk 1.4.41-rc1 Released.
+
+2011-02-21 14:57 +0000 [r308413]  Matthew Nicholson <mnicholson at digium.com>
+
+	* main/udptl.c: Properly check the bounds of arrays when decoding
+	  UDPTL packets. Also, remove broken support for receiving UDPTL
+	  packets larger than 16k. That shouldn't ever happen anyway.
+	  AST-2011-002 FAX-281
+
+2011-02-15 23:32 +0000 [r308002]  Jason Parker <jparker at digium.com>
+
+	* apps/app_queue.c: Fix regression that changed behavior of queues
+	  when ringing a queue member. This reverts r298596, which was to
+	  fix a highly bizarre and contrived issue with a queue member that
+	  called into his own queue being transferred back into his own
+	  queue. I couldn't reproduce that issue in any way. I think one of
+	  the other recent transfer fixes actually fixed this. (closes
+	  issue #18747) Reported by: vrban
+
+2011-02-11 00:29 +0000 [r307623]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Reentrancy problem if outgoing call gets
+	  different B channel than requested. The chan_dahdi
+	  pri_fixup_principle() routine needs to protect the Asterisk
+	  channel with the channel lock when it changes the technology
+	  private pointer to a new private structure. * Added lock
+	  protection while pri_fixup_principle() moves a call from one
+	  private structure to another. * Made some pri_fixup_principle()
+	  messages more meaningful. Partial backport from v1.8 -r300714.
+
+2011-02-10 22:33 +0000 [r307534]  Jason Parker <jparker at digium.com>
+
+	* main/asterisk.c: Remove color when executing commands via a
+	  remote console. Essentially this makes '-x' imply '-n' on
+	  rasterisk. This was done in a different and incomplete way
+	  previously, which I'll be reverting shortly. (issue #18776)
+	  Reported by: alecdavis
+
+2011-02-08 20:05 +0000 [r306972]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_sip.c: Fix comparison for REFER Replaces tags with
+	  pedantic=yes
+
+2011-02-08 19:40 +0000 [r306864-306965]  Jeff Peeler <jpeeler at digium.com>
+
+	* apps/app_voicemail.c: fix this line again
+
+	* apps/app_voicemail.c: clean this up, sorry my brain is not really
+	  working
+
+	* apps/app_voicemail.c: Backup file storing message duration is not
+	  used with IMAP_STORAGE, remove code. The message duration is
+	  stored in the body of the email when using IMAP_STORAGE, so
+	  nothing needs to happen with the backup file. (closes issue
+	  #18718) Reported by: kerframil
+
+	* apps/app_voicemail.c: make this safer and fully correct, pointed
+	  out by Steve Davis
+
+2011-02-07 22:35 +0000 [r306617-306672]  Terry Wilson <twilson at digium.com>
+
+	* res/res_features.c: Don't try to pickup a call in the middle of a
+	  masquerade If A calls B which doesn't answer and C & D both try
+	  to do a call pickup, it is possible for ast_pickup_call to answer
+	  the call, then fail to masquerade one of the calls because the
+	  other one is already in the process of masquerading. This patch
+	  checks to see if the channel is in the process of masquerading
+	  before call before selecting it for a pickup. Review:
+	  https://reviewboard.asterisk.org/r/1094/
+
+	* channels/chan_sip.c: Don't allow a REFER w/replaces to replace
+	  its own dialog Asterisk currently accepts a REFER with a Refer-To
+	  with an embedded Replaces header that matches the dialog of the
+	  REFER. This would be a situation like A calls B, A calls C, A
+	  transfers B to A, which is just silly. This patch makes the
+	  transfer fail instead of making Asterisk freak out and forget to
+	  hang other channels up. Review:
+	  https://reviewboard.asterisk.org/r/1093/
+
+2011-02-03 20:43 +0000 [r306120]  Jeff Peeler <jpeeler at digium.com>
+
+	* res/res_features.c: Fix no MOH and frame queueing problem for
+	  parked calls. This was a regression introduced when select was
+	  changed to poll and was just a conversion error: POLLPRI detects
+	  OOB data, not POLLERR. (closes issue #18637) Reported by: jvandal
+
+2011-02-03 20:36 +0000 [r306119]  Terry Wilson <twilson at digium.com>
+
+	* channels/chan_local.c: Set hangup cause in local_hangup When a
+	  call involves a local channel (like SIP -> Local -> SIP), the
+	  hangup cause was not being set. This resulted in SIP channels
+	  sometimes getting a 503 error instead of a 486 when the far side
+	  sent a busy. In Asterisk 1.8+ this also can cause issues with
+	  CCSS that involve a local channel. This patch sets the
+	  hangupcause for one side of the local channel to the other in
+	  local_hangup for outbound calls.
+
+2011-02-03 00:02 +0000 [r305888]  Richard Mudgett <rmudgett at digium.com>
+
+	* main/channel.c, channels/chan_sip.c, apps/app_sendtext.c: Minor
+	  AST_FRAME_TEXT related issues. * Include the null terminator in
+	  the buffer length. When the frame is queued it is copied. If the
+	  null terminator is not part of the frame buffer length, the
+	  receiver could see garbage appended onto it. * Add channel lock
+	  protection with ast_sendtext().
+
+2011-02-01 17:00 +0000 [r305471]  Jason Parker <jparker at digium.com>
+
+	* res/res_musiconhold.c: Close file descriptor for timing source
+	  when a MOH class gets destroyed. (closes issue #18457) Reported
+	  by: mcallist Patches: 18457-closetimer.diff uploaded by qwell
+	  (license 4) 18457-closetimer_trunk.diff uploaded by qwell
+	  (license 4) Tested by: qwell, loloski
+
+2011-01-31 23:45 +0000 [r305341]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c: Obtain the pri lock for PRI queue
+	  counters. Need to obtain the pri lock when calling
+	  pri_dump_info_str() to avoid a reentrancy problem when
+	  calculating the Q.921 Q count statistic. JIRA AST-484
+
+2011-01-31 22:56 +0000 [r305129-305252]  Jason Parker <jparker at digium.com>
+
+	* apps/app_dial.c, channels/chan_sip.c: Prevent a crash when
+	  dialing a technology with no destination (ex: Dial(SIP/))
+	  chan_iax2 and other channel drivers already had code to prevent
+	  this. The attempt that app_dial was making to prevent it was not
+	  correct, so I fixed that. (closes issue #18371) Reported by:
+	  gbour Patches: 18371.patch uploaded by gbour (license 1162)
+
+	* res/res_musiconhold.c: Set file descriptors to -1 on creation, so
+	  that we don't see weirdness later.
+
+2011-01-31 06:54 +0000 [r304952]  Tilghman Lesher <tilghman at meg.abyt.es>
+
+	* apps/app_voicemail.c: Fix compilation when ODBC_STORAGE is
+	  defined.
+
+2011-01-29 21:48 +0000 [r304820]  Sean Bright <sean at malleable.com>
+
+	* apps/app_meetme.c: Backport MeetMe related reference leaks fixes
+	  from 1.6.2/1.8/trunk. I had forgotten that MeetMe in 1.4 also
+	  used astobj2, so backport the fixes where appropriate.
+
+2011-01-27 16:57 +0000 [r304460-304464]  Jason Parker <jparker at digium.com>
+
+	* configure, configure.ac: Fix default prefix=/usr regression on
+	  non-Linux systems. This partially reverts a change made in
+	  branches/1.4/ r267759, which will cause issue #17013 to be
+	  reopened. This issue was pointed out by a user on #asterisk, who
+	  helpfully discovered that paths were being set incorrectly. To
+	  truly understand what was wrong, one should run: svn diff --force
+	  -c<this revision> configure
+
+	* configure, include/asterisk/autoconfig.h.in: Rerun bootstrap.sh
+	  with no changes, so that it is more obvious what my next commit
+	  changes.
+
+2011-01-26 21:00 +0000 [r304247]  Matthew Nicholson <mnicholson at digium.com>
+
+	* channels/chan_sip.c: Convert from network to host byte ordering
+	  before checking if an IP is a multicast address.
+
+2011-01-26 20:38 +0000 [r304242]  Mark Michelson <mmichelson at digium.com>
+
+	* main/udptl.c: Get rid of unused 'verbose' field in ast_udptl
+
+2011-01-26 20:38 +0000 [r304241]  Matthew Nicholson <mnicholson at digium.com>
+

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