[asterisk-commits] russell: tag 1.4.19-rc1 r105670 - /tags/1.4.19-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Mar 4 11:54:53 CST 2008


Author: russell
Date: Tue Mar  4 11:54:52 2008
New Revision: 105670

URL: http://svn.digium.com/view/asterisk?view=rev&rev=105670
Log:
Importing files for 1.4.19-rc1 release

Added:
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    tags/1.4.19-rc1/.version   (with props)
    tags/1.4.19-rc1/ChangeLog   (with props)

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--- tags/1.4.19-rc1/ChangeLog (added)
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+2008-03-04  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.19-rc1 released.
+
+2008-03-04 04:31 +0000 [r105591]  Russell Bryant <russell at digium.com>
+
+	* main/pbx.c: Backport a minor bug fix from trunk that I found
+	  while doing random code cleanup. Properly break out of the loop
+	  when a context isn't found when verify that includes are valid.
+
+2008-03-03 18:06 +0000 [r105572]  Jason Parker <jparker at digium.com>
+
+	* res/snmp/agent.c: Fix type for astNumChannels. (closes issue
+	  #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
+	  (license 192)
+
+2008-03-03 17:16 +0000 [r105563-105570]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_local.c: In the case of an ast_channel allocation
+	  failure, take the local_pvt out of the pvt list before destroying
+	  it.
+
+	* channels/chan_local.c: Fix a potential memory leak of the
+	  local_pvt struct when ast_channel allocation fails. Also, in
+	  passing, centralize the code necessary to destroy a local_pvt.
+
+	* main/autoservice.c: Update the copyright information for
+	  autoservice. Most of the code in this file now is stuff that I
+	  have written recently ...
+
+	* main/asterisk.c, main/channel.c, include/asterisk.h,
+	  main/autoservice.c: Merge in some changes from
+	  team/russell/autoservice-nochans-1.4 These changes fix up some
+	  dubious code that I came across while auditing what happens in
+	  the autoservice thread when there are no channels currently in
+	  autoservice. 1) Change it so that autoservice thread doesn't keep
+	  looping around calling ast_waitfor_n() on 0 channels twice a
+	  second. Instead, use a thread condition so that the thread
+	  properly goes to sleep and does not wake up until a channel is
+	  put into autoservice. This actually fixes an interesting bug, as
+	  well. If the autoservice thread is already running (almost always
+	  is the case), then when the thread goes from having 0 channels to
+	  have 1 channel to autoservice, that channel would have to wait
+	  for up to 1/2 of a second to have the first frame read from it.
+	  2) Fix up the code in ast_waitfor_nandfds() for when it gets
+	  called with no channels and no fds to poll() on, such as was the
+	  case with the previous code for the autoservice thread. In this
+	  case, the code would call alloca(0), and pass the result as the
+	  first argument to poll(). In this case, the 2nd argument to
+	  poll() specified that there were no fds, so this invalid pointer
+	  shouldn't actually get dereferenced, but, this code makes it
+	  explicit and ensures the pointers are NULL unless we have valid
+	  data to put there. (related to issue #12116)
+
+2008-03-03 15:28 +0000 [r105557-105560]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c: It is possible for no audio to pass between the
+	  current digit and next digit so expand logic that clears
+	  emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
+	  edgreenberg Patches: v1-11911.patch uploaded by dimas (license
+	  88) Tested by: tbsky
+
+	* channels/chan_sip.c: Add a comment to describe some logic.
+	  (closes issue #12120) Reported by: flefoll Patches:
+	  chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
+	  244)
+
+2008-02-29 23:34 +0000 [r105409]  Russell Bryant <russell at digium.com>
+
+	* main/autoservice.c: Fix a major bug in autoservice. There was a
+	  race condition in the handling of the list of channels in
+	  autoservice. The problem was that it was possible for a channel
+	  to get removed from autoservice and destroyed, while the
+	  autoservice thread was still messing with the channel. This led
+	  to memory corruption, and caused crashes. This explains multiple
+	  backtraces I have seen that have references to autoservice, but
+	  do to the nature of the issue (memory corruption), could cause
+	  crashes in a number of areas. (fixes the crash in BE-386) (closes
+	  issue #11694) (closes issue #11940) The following issues could be
+	  related. If you are the reporter of one of these, please update
+	  to include this fix and try again. (potentially fixes issue
+	  #11189) (potentially fixes issue #12107) (potentially fixes issue
+	  #11573) (potentially fixes issue #12008) (potentially fixes issue
+	  #11189) (potentially fixes issue #11993) (potentially fixes issue
+	  #11791)
+
+2008-02-29 14:47 +0000 [r105326]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* res/res_jabber.c: Fix a potential memory leak
+
+2008-02-29 14:34 +0000 [r105296]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: If the message file does not exist, just
+	  return harmlessly, instead of crashing. (Closes issue #12108)
+
+2008-02-29 13:48 +0000 [r105261]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_voicemail.c: Bump up the size of the uniqueid variable.
+	  (closes issue #12107) Reported by: asgaroth
+
+2008-02-29 13:05 +0000 [r105209]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* res/res_jabber.c: Automatically create new buddy upon reception
+	  of a presence stanza of type subscribed. (closes issue #12066)
+	  Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
+	  phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
+	  (license 73) Tested by: ffadaie, phsultan
+
+2008-02-28 22:23 +0000 [r105116]  Russell Bryant <russell at digium.com>
+
+	* main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
+	  tracking code that was discovered by mmichelson. The issue is
+	  that if the lock history array was full, then the functions to
+	  mark a lock as acquired or not would adjust the stats for
+	  whatever lock is at the end of the array, which may not be
+	  itself. So, do a sanity check to make sure that we're updating
+	  lock info for the proper lock. (This explains the bizarre stats
+	  on lock #63 in BE-396, thanks Mark!)
+
+2008-02-28 21:56 +0000 [r105113]  Tilghman Lesher <tlesher at digium.com>
+
+	* contrib/init.d/rc.debian.asterisk: Update init script for LSB
+	  compat (closes issue #9843) Reported by: ibc Patches:
+	  rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
+	  paravoid
+
+2008-02-28 20:11 +0000 [r105059]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: When using autofill, members who are in use
+	  should be counted towards the number of available members to call
+	  if ringinuse is set to yes. Thanks to jmls who brought this issue
+	  up on IRC
+
+2008-02-28 19:20 +0000 [r104920-105005]  Jason Parker <jparker at digium.com>
+
+	* main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
+	  applications, if we get NULL. This protects against possible
+	  segfaults in applications that may try to use data before
+	  checking length (ast_strdupa'ing it, for example) (closes issue
+	  #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
+	  uploaded by qwell (license 4)
+
+	* channels/chan_skinny.c: According to a video at www.cisco.com,
+	  the 7921G supports 6 line appearances.
+
+2008-02-28 00:05 +0000 [r104868]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/Makefile, build_tools/strip_nonapi: Compatibility fix for
+	  PPC64 (closes issue #12081) Reported by: jcollie Patches:
+	  asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
+	  Tested by: jcollie, Corydon76
+
+2008-02-27 21:49 +0000 [r104841]  Mark Michelson <mmichelson at digium.com>
+
+	* main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
+	  lockable list. This is because in some cases, the ast_dial may
+	  exist in multiple threads due to asynchronous execution of its
+	  application, and I found some cases where race conditions could
+	  exist. 2. Check in ast_dial_join to be sure that the channel
+	  still exists before attempting to lock it, since it could have
+	  gotten hung up but the is_running_app flag on the
+	  ast_dial_channel may not have been cleared yet. (closes issue
+	  #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
+	  putnopvut (license 60) Tested by: jvandal
+
+2008-02-27 20:56 +0000 [r104787]  Joshua Colp <jcolp at digium.com>
+
+	* apps/app_chanspy.c: Don't loop around infinitely trying to spy on
+	  our own channel, and don't forget to free/detach the datastore
+	  upon hangup of the spy.
+
+2008-02-27 20:36 +0000 [r104783]  Mark Michelson <mmichelson at digium.com>
+
+	* main/file.c: Bump a couple of more buffers up by 2 so that
+	  annoying warnings aren't generated like crazy on every
+	  fileexists_core call.
+
+2008-02-27 18:15 +0000 [r104704]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/manager.c: Ensure the session ID can't be 0.
+
+2008-02-27 17:41 +0000 [r104665]  Joshua Colp <jcolp at digium.com>
+
+	* main/file.c: Bump up the buffer by 2.
+
+2008-02-27 17:33 +0000 [r104625]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
+	  stuck in an infinite loop without being able to detect that the
+	  calling channel hung up. (closes issue #12076, reported by junky,
+	  patched by me)
+
+2008-02-27 17:26 +0000 [r104598]  Jason Parker <jparker at digium.com>
+
+	* res/res_features.c: Inherit language from the transfering channel
+	  on a blind transfer. (closes issue #11682) Reported by: caio1982
+	  Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
+	  (license 22) Tested by: caio1982, victoryure
+
+2008-02-27 17:07 +0000 [r104596]  Joshua Colp <jcolp at digium.com>
+
+	* main/loader.c: Use the lock (which already existed, it just
+	  wasn't used) on the updaters list to protect the contents instead
+	  of the overall module list lock. (closes issue #12080) Reported
+	  by: ChaseVenters
+
+2008-02-27 16:53 +0000 [r104593]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/file.c: fallback to standard English prompts properly when
+	  using new prompt directory layout (closes issue #11831) Reported
+	  by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
+	  20) (modified by me to improve code and conform rest of function
+	  to coding guidelines)
+
+2008-02-27 16:45 +0000 [r104591]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: When we receive a known alarm, make sure
+	  that the unknown alarm flag is not still set to make sure that
+	  when we come back out of alarm, it gets reported in the log and
+	  manager interface (after discussion with tzafrir on the -dev
+	  list)
+
+2008-02-27 15:52 +0000 [r104536]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_smdi.c: Only stop the MWI monitor thread if it was
+	  actually started. (closes issue #12086) Reported by: francesco_r
+
+2008-02-27 01:15 +0000 [r104332-104334]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c: Avoid some recursion in the cleanup code for
+	  the chanspy datastore (closes issue #12076, reported by junky,
+	  patched by me)
+
+	* channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
+	  an alarm. However, Asterisk 1.4 does not know what to do with
+	  these alarms. Only Asterisk 1.6 cares about it. So, if we get an
+	  unknown alarm in chan_zap, don't generate confusing log messages
+	  about it.
+
+2008-02-26 18:26 +0000 [r104132-104141]  Jason Parker <jparker at digium.com>
+
+	* Makefile: Add badshell to .PHONY target (thanks Kevin)
+
+	* Makefile: Since all shells aren't as awesome as bash, we have to
+	  fail if somebody tries to use a literal "~" in DESTDIR.
+
+	* sounds/Makefile: Revert previous abspath change. ...abspath is
+	  new in GNU make 3.81. I feel so...defeated. Must find new fix!
+
+	* sounds/Makefile: Fix a very bizarre issue we were seeing with our
+	  buildbot when using a DESTDIR that wasn't an absolute path (such
+	  as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
+	  that some of the targets were being expanded to the full path, so
+	  $@ ended up being /root/asterisk-1.4/[...]/ rather than
+	  ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
+	  in GNU make. (*cough*
+	  http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
+
+2008-02-26 00:25 +0000 [r104119]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/smdi.h, apps/app_voicemail.c,
+	  channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
+	  Merge changes from team/russell/smdi-1.4 This commit brings in a
+	  significant set of changes to the SMDI support in Asterisk. There
+	  were a number of bugs in the current implementation, most notably
+	  being that it was very likely on busy systems to pop off the
+	  wrong message from the SMDI message queue. So, this set of
+	  changes fixes the issues discovered as well as introducing some
+	  new ways to use the SMDI support which are required to avoid the
+	  bugs with grabbing the wrong message off of the queue. This code
+	  introduces a new interface to SMDI, with two dialplan functions.
+	  First, you get an SMDI message in the dialplan using
+	  SMDI_MSG_RETRIEVE() and then you access details in the message
+	  using the SMDI_MSG() function. A side benefit of this is that it
+	  now supports more than just chan_zap. For example, with this
+	  implementation, you can have some FXO lines being terminated on a
+	  SIP gateway, but the SMDI link in Asterisk. Another issue with
+	  the current implementation is that it is quite common that the
+	  station ID that comes in on the SMDI link is not necessarily the
+	  same as the Asterisk voicemail box. There are now additional
+	  directives in the smdi.conf configuration file which let you map
+	  SMDI station IDs to Asterisk voicemail boxes. Yet another issue
+	  with the current SMDI support was related to MWI reporting over
+	  the SMDI link. The current code could only report a MWI change
+	  when the change was made by someone calling into voicemail. If
+	  the change was made by some other entity (such as with IMAP
+	  storage, or with a web interface of some kind), then the MWI
+	  change would never be sent. The SMDI module can now poll for MWI
+	  changes if configured to do so. This work was inspired by and
+	  primarily done for the University of Pennsylvania. (also related
+	  to issue #9260)
+
+2008-02-26 00:03 +0000 [r104111]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
+	  (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
+	  uploaded by putnopvut (license 60)
+
+2008-02-25 23:42 +0000 [r104102-104106]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c: This patch fixes some pretty significant
+	  problems with how app_chanspy handles pointers to channels that
+	  are being spied upon. It was very likely that a crash would occur
+	  if the channel being spied upon hung up. This was because the
+	  current ast_channel handling _requires_ that the object is locked
+	  or else it could disappear at any time (except in the owning
+	  channel thread). So, this patch uses some channel datastore magic
+	  on the spied upon channel to be able to detect if and when the
+	  channel goes away. (closes issue #11877) (patch written by me,
+	  but thanks to kpfleming for the idea, and to file for review)
+
+	* main/utils.c: Improve the lock tracking code a bit so that a
+	  bunch of old locks that threads failed to lock don't sit around
+	  in the history. When a lock is first locked, this checks to see
+	  if the last lock in the list was one that was failed to be
+	  locked. If it is, then that was a lock that we're no longer
+	  sitting in a trylock loop trying to lock, so just remove it.
+	  (inspired by issue #11712)
+
+2008-02-25 21:37 +0000 [r104095]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Make it so a users.conf user creates both a
+	  SIP peer and a SIP user. The user will be used for inbound
+	  authentication for the device, and peer will be used for placing
+	  calls to the device. (closes issue #9044) Reported by: queuetue
+	  Patches: sip-gui-friend.diff uploaded by qwell (license 4)
+
+2008-02-25 21:31 +0000 [r104094]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: If the destination folder is full, don't
+	  delete a message when exiting. (closes issue #12065) Reported by:
+	  selsky Patch by: (myself)
+
+2008-02-25 20:49 +0000 [r104092]  Jason Parker <jparker at digium.com>
+
+	* main/config.c: Allow the use of #include and #exec in situations
+	  where the max include depth was only 1. Specifically, this fixes
+	  using #include and #exec in extconfig.conf. This was basically
+	  caused because the config file itself raises the include level to
+	  1. I opted not to raise the include limit, because recursion here
+	  could cause very bizarre behavior. Pointed out, and tested by
+	  jmls (closes issue #12064)
+
+2008-02-25 18:38 +0000 [r104086]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_agent.c: Ensure that the channel doesn't disappear
+	  in agent_logoff(). If it does, it could cause a crash. (fixes the
+	  crash reported in BE-396)
+
+2008-02-25 16:16 +0000 [r104082-104084]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: If a resubscription comes in for a dialog we
+	  no longer know about tell the remote side that the dialog does
+	  not exist so they subscribe again using a new dialog. (closes
+	  issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
+	  uploaded by file (license 11)
+
+	* channels/chan_sip.c: Due to recent changes tag will no longer be
+	  NULL if not present so we have to use ast_strlen_zero to see if
+	  it's actually blank. (closes issue #12061) Reported by: flefoll
+	  Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
+	  flefoll (license 244)
+
+2008-02-22 22:45 +0000 [r104037]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_sip.c: Backwards debug message. (closes issue
+	  #12052) Reported by: flefoll Patches:
+	  chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
+	  244)
+
+2008-02-21 21:05 +0000 [r104026-104027]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_zap.c: And as a followup to revision 104026,
+	  completely remove event-related calls from a section of code
+	  where we know there was no event to handle or get.
+
+	* channels/chan_zap.c: Remove an incorrect debug message. It
+	  reported that it had received a specific event and tried to
+	  report which event was received. What actually was happening was
+	  that it was reporting the number of bytes returned from a call to
+	  read(). Thanks to Jared Smith for bringing the issue up on IRC
+
+2008-02-21 14:33 +0000 [r104015]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/manager.c: reduce the likelihood that HTTP Manager session
+	  ids will consist of primarily '1' bits
+
+2008-02-20 22:32 +0000 [r103956]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Clear up confusion when viewing the
+	  QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
+	  user's perspective, the queue does exist, we shouldn't tell them
+	  we couldn't find the queue. Instead since it is a dead queue,
+	  report a 0 waiting count This issue was brought up on IRC by jmls
+
+2008-02-20 22:06 +0000 [r103953]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_zap.c: Don't wait for additional digits when
+	  overlap dialing is enabled if the setup message contains the
+	  sending_complete information element. (closes issue #11785)
+	  Reported by: klaus3000 Patches:
+	  sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
+	  klaus3000 (license 65)
+
+2008-02-20 21:40 +0000 [r103904]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_local.c: Fix a crash if the channel becomes NULL
+	  while attempting to lock it. (closes issue #12039) Reported by:
+	  danpwi
+
+2008-02-20 17:53 +0000 [r103845]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/stdtime/localtime.c: Compat fix for Solaris (closes issue
+	  #12022) Reported by: asgaroth Patches:
+	  20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
+	  Tested by: asgaroth
+
+2008-02-19 20:28 +0000 [r103823]  Joshua Colp <jcolp at digium.com>
+
+	* channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
+	  (closes issue #11241) Reported by: tusar Patches:
+	  h323id_as_callerid_name.patch uploaded by tusar (license 344)
+
+2008-02-19 20:02 +0000 [r103821]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_local.c: Account for the fact that the "other"
+	  channel can disappear while the local pvt is not locked. (fixes a
+	  problem introduced in rev 100581) (closes issue #12012) Reported
+	  by: stevedavies Patch by me
+
+2008-02-19 17:31 +0000 [r103807-103812]  Joshua Colp <jcolp at digium.com>
+
+	* configure, configure.ac: Don't look for launchd when cross
+	  compiling. (closes issue #12029) Reported by: ovi
+
+	* channels/chan_sip.c: Fix building of chan_sip.
+
+2008-02-19 10:27 +0000 [r103806]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Make sure we send error replies correctly by
+	  checking the via header.
+
+2008-02-18 23:56 +0000 [r103801]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c: Ensure that emulated DTMFs do not get interrupted
+	  by another begin frame. (closes issue #11740) Reported by: gserra
+	  Patches: v1-11740.patch uploaded by dimas (license 88) (closes
+	  issue #11955) Reported by: tsearle (closes issue #10530) Reported
+	  by: xmarksthespot
+
+2008-02-18 22:28 +0000 [r103790-103795]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_zap.c: Fix previous commit so that we actually
+	  disable echocanbridged if echocancel is off.
+
+	* channels/chan_zap.c: Correct a message when echocancelwhenbridged
+	  is on, but echocancel is not. Issue #12019
+
+2008-02-18 20:52 +0000 [r103786]  Mark Michelson <mmichelson at digium.com>
+
+	* main/app.c: There was an invalid assumption when calculating the
+	  duration of a file that the filestream in question was created
+	  properly. Unfortunately this led to a segfault in the situation
+	  where an unknown format was specified in voicemail.conf and a
+	  voicemail was recorded. Now, we first check to be sure that the
+	  stream was written correctly or else assume a zero duration.
+	  (closes issue #12021) Reported by: jakep Tested by: putnopvut
+
+2008-02-18 17:31 +0000 [r103780]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/rtp.c, channels/chan_sip.c: When a SIP channel is being
+	  auto-destroyed, it's possible for it to still be in bridge code.
+	  When that happens, we crash. Delay the RTP destruction until the
+	  bridge is ended. (closes issue #11960) Reported by: norman
+	  Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
+	  (license 14) Tested by: norman
+
+2008-02-18 16:37 +0000 [r103770]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_zap.c: Fix a linked list corruption that under the
+	  right circumstances could lead to a looped list, meaning it will
+	  traverse forever. (closes issue #11818) Reported by: michael-fig
+	  Patches: 11818.patch uploaded by putnopvut (license 60) Tested
+	  by: michael-fig
+
+2008-02-18 16:11 +0000 [r103763-103768]  Joshua Colp <jcolp at digium.com>
+
+	* main/asterisk.c: Backport fix from issue #9325. (closes issue
+	  #11980) Reported by: rbrunka
+
+	* channels/chan_sip.c: Don't care if the extension given doesn't
+	  exist for subscription based MWI.
+
+2008-02-15 23:31 +0000 [r103726-103741]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
+	  condition (closes issue #11780) Reported by: guillecabeza
+	  Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
+	  380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
+	  380)
+
+	* main/loader.c: In the case that you try to directly reload a
+	  module has returned AST_MODULE_LOAD_DECLINE, log a message
+	  indicating that the module is not fully initialized and must be
+	  initialized using "module load".
+
+	* main/loader.c: Don't attempt to execute the reload callback for a
+	  module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
+	  that was reported against chan_console in trunk. (closes issue
+	  #11953, reported by junky, fixed by me)
+
+2008-02-15 17:26 +0000 [r103688-103722]  Mark Michelson <mmichelson at digium.com>
+
+	* doc/imapstorage.txt, configure, configure.ac: Final round of
+	  changes for configure script logic for IMAP Now if a directory is
+	  specified, then we will search that directory for a source
+	  installation of the IMAP toolkit. If none is found, then we will
+	  use that directory as the basis for detecting a package
+	  installation of the IMAP c-client. If that check fails, then
+	  configure will fail.
+
+	* configure, configure.ac: Fix a bit of wrong logic in the
+	  configure script that caused problems when trying to configure
+	  without IMAP. Patch suggestion from phsultan, but I modified it
+	  slightly. (closes issue #12003) Reported by: pj Tested by:
+	  putnopvut
+
+	* doc/imapstorage.txt, configure, configure.ac: I apparently
+	  misunderstood one of the requirements of this configure change.
+	  Now, if a source directory is specified with the --with-imap
+	  option, and a valid source installation is not detected there,
+	  then configure will fail and will not check for a package
+	  installation.
+
+	* doc/imapstorage.txt: Make a small clarification in the
+	  documentation
+
+	* doc/imapstorage.txt: Update documentation regarding configuration
+	  of IMAP
+
+	* apps/app_voicemail.c, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Change to the
+	  configure logic regarding IMAP. Prior to this commit, if you
+	  wished to configure Asterisk with IMAP support, you would use the
+	  --with-imap configure switch in one of the following two ways:
+	  --with-imap=/some/directory would look in the directory specified
+	  for a UW IMAP source installation --with-imap would assume that
+	  you had imap-2004g installed in .. relative to the Asterisk
+	  source With this set of changes the two above options still work
+	  the same, but there are two new behaviors, too.
+	  --with-imap=system will assume that you have -libc-client.so
+	  where you store your shared objects and will attempt to find
+	  c-client headers in your include path either in the imap or
+	  c-client directory. If either of the two original methods of
+	  specifying the imap option should fail, then the check for
+	  --with-imap =system will be performed in addition. It is only
+	  after this "system" check that failure can happen.
+
+	* apps/app_voicemail.c: Fix build for non-IMAP builds
+
+	* apps/app_voicemail.c: Fix the new message count if delete=yes
+	  when using IMAP storage. (closes issue #11406) Reported by:
+	  jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
+	  50) Tested by: jaroth
+
+2008-02-14 19:51 +0000 [r103683-103684]  Jason Parker <jparker at digium.com>
+
+	* funcs/func_cdr.c: swap location for this..
+
+	* funcs/func_cdr.c: Document the 'l' option to the CDR() function.
+	  (Thanks voipgate for pointing out the option, and Leif for
+	  providing text for it.) Closes issue #11695.
+
+2008-02-13 06:25 +0000 [r103556-103607]  Tilghman Lesher <tlesher at digium.com>
+
+	* channels/chan_agent.c: We aren't talking to ourselves; we're
+	  talking to someone else. (closes issue #11771) Reported by:
+	  msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
+	  (license 22) Tested by: caio1982, msetim
+
+	* apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
+	  not loaded (which is a symbol dependency) (closes issue #11760)
+	  Reported by: non-poster Patches: 20080114__bug11760.diff.txt
+	  uploaded by Corydon76 (license 14) Tested by: Corydon76,
+	  non-poster, jamesgolovich
+
+2008-02-12 22:24 +0000 [r103503-103504]  Jason Parker <jparker at digium.com>
+
+	* main/asterisk.c: revert accidental change from last commit. oops
+
+	* contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
+	  that was impossible.
+
+2008-02-12 15:09 +0000 [r103324-103385]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Even if no CallerID name or number has been
+	  provided by the remote party still use the configured sip.conf
+	  ones. (closes issue #11977) Reported by: pj
+
+	* apps/app_meetme.c: If entering a conference with the 'w' option
+	  ensure that we can't listen or speak until the marked user
+	  appears. (closes issue #11835) Reported by: alanmcmillan
+
+2008-02-11 17:05 +0000 [r103315]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/zapata.conf.sample: improve 2BCT documentation a bit
+	  (thanks Jared)
+
+2008-02-09 06:23 +0000 [r103197]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Commit fix for being unable to send
+	  voicemail from VoiceMailMain Reported by: William F Acker (via
+	  the -users mailing list) Patch by: Corydon76 (license 14)
+
+2008-02-08 18:48 +0000 [r103070-103120]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Prevent a potential three-thread deadlock. Also
+	  added a comment block to explicitly state the locking order
+	  necessary inside app_queue. (closes issue #11862) Reported by:
+	  flujan Patches: 11862.patch uploaded by putnopvut (license 60)
+	  Tested by: flujan
+
+	* channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
+	  fails for Zaptel timing device. (closes issue #11891) Reported
+	  by: tzafrir
+
+2008-02-08 15:08 +0000 [r102968]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_iax2.c: Make sure the presence of dbsecret is
+	  factored into user scoring. (closes issue #11952) Reported by:
+	  bbhoss
+
+2008-02-07 19:53 +0000 [r102858]  Jason Parker <jparker at digium.com>
+
+	* res/res_features.c: Specify which digit string was matched in
+	  debug message. (closes issue #11949) Reported by: dimas Patches:
+	  v1-feature-debug.patch uploaded by dimas (license 88)
+
+2008-02-07 16:41 +0000 [r102807]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* configs/zapata.conf.sample: document usage of 'transfer'
+	  configuration option for ISDN PRI switch-side transfers
+
+2008-02-06 17:59 +0000 [r102653-102725]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: Only consider a T.38-only INVITE compatible
+	  if we have both a joint capability between us and them and if
+	  they provided T.38.
+
+	* main/global_datastores.c: Add missing header file and
+	  ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
+	  snuffy
+
+2008-02-06 15:19 +0000 [r102651]  Russell Bryant <russell at digium.com>
+
+	* configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
+	  that it gets inherited by outbound channels. (due to a discussion
+	  between me and a user via email)
+
+2008-02-06 11:48 +0000 [r102627]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* pbx/Makefile, res/Makefile: ensure that all remaining
+	  multi-object modules are built using their proper CFLAGS and
+	  include directory paths
+
+2008-02-06 00:26 +0000 [r102576]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Move around some defines to unbreak ODBC
+	  storage. (closes issue #11932) Reported by: snuffy
+
+2008-02-05 20:02 +0000 [r102453]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
+	  issue #11919) Reported by: eferro Patches:
+	  mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
+	  Tested by: eferro
+
+2008-02-05 19:52 +0000 [r102450]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: If a REGISTER attempt comes in that is a
+	  retransmission of a previous REGISTER do not create a new nonce
+	  value. (issue #BE-381)
+
+2008-02-05 17:15 +0000 [r102425]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/Makefile: ensure that components of chan_misdn.so are
+	  built using any special build options that the configure script
+	  generated (reported by Philipp Kempgen on asterisk-dev)
+
+2008-02-05 15:09 +0000 [r102378]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_clioriginate.c: Perform dialing asynchronously when using
+	  the originate CLI command so the CLI does not appear to block.
+	  (closes issue #11927) Reported by: bbhoss
+
+2008-02-04 21:06 +0000 [r102214-102323]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/asterisk.c, utils/muted.c, configure,
+	  include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
+	  fix: OS X now deprecates the use of the daemon(3) API. (closes
+	  issue #11908) Reported by: oej Patches:
+	  20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
+	  Tested by: Corydon76
+
+	* funcs/func_strings.c: Missing braces. (closes issue #11912)
+	  Reported by: dimas Patches: sprintf.patch uploaded by dimas
+	  (license 88)
+
+2008-02-03 16:38 +0000 [r102090-102142]  Olle Johansson <oej at edvina.net>
+
+	* channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
+	  (according to RFC 3261) (closes issue #9492) Reported by:
+	  kryptolus Patches: bug9492.txt uploaded by oej (license 306)
+	  Tested by: oej
+
+	* channels/chan_sip.c: Handle ACK and CANCEL in an invite
+	  transaction - even if we get INFO transactions during the actual
+	  call setup. (closes issue #10567) Reported by: jacksch Tested by:
+	  oej Patch by: oej inspired by suggestions from neutrino88 in the
+	  bug tracker
+
+2008-02-01 23:06 +0000 [r101989]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
+	  out that even though G.722 is 16 kHz, it is supposed to specified
+	  as 8 kHz in the RTP, and RTP timestamps are supposed to be
+	  calculated based on 8 kHz. (Apparently this is due to a bug in a
+	  spec, but people follow it anyway, because it's the spec ...)
+
+2008-02-01 21:54 +0000 [r101894-101942]  Tilghman Lesher <tlesher at digium.com>
+
+	* apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
+	  voicemail, and fixed several other bugs while I'm in the area.
+	  (closes issue #11615) Reported by: jamessan Patches:
+	  20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
+	  Tested by: Corydon76, jamessan
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  acinclude.m4: Change detection of getifaddrs to use
+	  AST_C_COMPILE_CHECK, backported from trunk (as suggested by
+	  kpfleming)
+
+2008-02-01 17:41 +0000 [r101822]  Jason Parker <jparker at digium.com>
+
+	* apps/app_authenticate.c: Remove a needless (and incorrect) call
+	  to feof() after fgets(). This would have exited the loop early if
+	  you had an authentication file with no newline at the end.
+
+2008-02-01 17:27 +0000 [r101818-101820]  Russell Bryant <russell at digium.com>
+
+	* apps/app_authenticate.c: off by one error
+
+	* apps/app_authenticate.c: Don't overwrite the last character of a
+	  line if it's not a newline. This would happen if the last line in
+	  the file doesn't have a newline. (pointed out by Qwell)
+
+2008-02-01 15:55 +0000 [r101772]  Tilghman Lesher <tlesher at digium.com>
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  main/acl.c: Compatibility fix for OpenWRT (reported by Brian
+	  Capouch via the mailing list)
+
+2008-02-01 00:32 +0000 [r101693]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
+	  strings for the case that no peer or hostname was provided, which
+	  is the one part of the dial string that is absolutely required.
+	  If it's not there, bail out. (closes issue #11897) Reported by
+	  sokhapkin Patch by me
+
+2008-02-01 00:06 +0000 [r101649]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
+	  periods of no channel activity" (closes issue #9256) Reported by:
+	  cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
+	  uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
+	  rjain
+
+2008-01-31  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.18 released.
+
+2008-01-31 23:10 +0000 [r101601]  Russell Bryant <russell at digium.com>
+
+	* main/translate.c, main/file.c: Fix a couple of places where
+	  ast_frfree() was not called on a frame that came from a
+	  translator. This showed itself by g729 decoders not getting
+	  released. Since the flag inside the translator frame never got
+	  unset by freeing the frame to indicate it was no longer in use,
+	  the translators never got destroyed, and thus the g729 licenses
+	  were not released. (closes issue #11892) Reported by: xrg
+	  Patches: 11892.diff uploaded by russell (license 2) Tested by:
+	  xrg, russell
+
+2008-01-31 21:00 +0000 [r101531]  Mark Michelson <mmichelson at digium.com>
+
+	* res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
+	  beginning of an absolute pathname. 2. If ast_monitor_change_fname
+	  is called and the new filename is the same as the old, then exit
+	  early and don't set the filename_changed field in the monitor
+	  structure. Setting it in this case was causing ast_monitor_stop
+	  to erroneously delete them. (closes issue #11741) Reported by:
+	  garlew Tested by: putnopvut
+
+2008-01-31 19:52 +0000 [r101482]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
+	  for struct in_addr funkiness. Issue #11885, patch by snuffy.
+
+2008-01-31 19:30 +0000 [r101480]  Steve Murphy <murf at digium.com>
+
+	* main/pbx.c: closes issue #11845; that's the one where there's a
+	  1004 byte cdr leak with every AMI Redirect to a zap channel
+
+2008-01-31 19:17 +0000 [r101413-101433]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_agent.c: Add more missing locking of the agents
+	  list ...
+
+	* channels/chan_agent.c: Move the locking from find_agent() into
+	  the agent dialplan function handler to ensure that the agent
+	  doesn't disappear while we're looking at it.
+
+	* channels/chan_agent.c: Add missing locking to the find_agent()
+	  function.
+

[... 15393 lines stripped ...]



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