[svn-commits] russell: tag 1.4.11 r80192 - in /tags/1.4.11: .lastclean .version ChangeLog

SVN commits to the Digium repositories svn-commits at lists.digium.com
Tue Aug 21 14:07:46 CDT 2007


Author: russell
Date: Tue Aug 21 14:07:45 2007
New Revision: 80192

URL: http://svn.digium.com/view/asterisk?view=rev&rev=80192
Log:
importing files for 1.4.11 release

Added:
    tags/1.4.11/.lastclean   (with props)
    tags/1.4.11/.version   (with props)
    tags/1.4.11/ChangeLog   (with props)

Added: tags/1.4.11/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.11/.lastclean?view=auto&rev=80192
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--- tags/1.4.11/ChangeLog (added)
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@@ -1,0 +1,10849 @@
+2007-08-21  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.11 released.
+
+2007-08-21 18:42 +0000 [r80183]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Don't record SIP dialog history if it's not
+	  turned on. Also, put an upper limit on how many history entires
+	  will be stored for each SIP dialog. It is currently set to 50,
+	  but can be increased if deemed necessary. (closes issue #10421,
+	  closes issue #10418, patches suggested by jmoldenhauer, patches
+	  updated by me) (Security implications documented in AST-2007-020)
+
+2007-08-21 16:39 +0000 [r80166-80167]  Steve Murphy <murf at digium.com>
+
+	* include/asterisk/alaw.h, include/asterisk/ulaw.h: ugh. removing
+	  the diffs from ulaw.h and alaw.h for now; accidentally added them
+	  in 80166
+
+	* main/alaw.c, include/asterisk/alaw.h, include/asterisk/ulaw.h:
+	  This patch solves problem 1 in 8126; it should not slow down the
+	  alaw codec, but should prevent signal degradation via multiple
+	  trips thru the codec. Fossil estimates the twice thru this codec
+	  will prevent fax from working. 4-6 times thru would result
+	  hearable, noticeable, voice degradation.
+
+2007-08-21 15:22 +0000 [r80132]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_mgcp.c: Don't try to dereference the owner channel
+	  when it may not exist (issue #10507, maxper)
+
+2007-08-21 15:03 +0000 [r80130]  Jason Parker <jparker at digium.com>
+
+	* configs/cdr.conf.sample: (issue #10510) Reported by: casper
+	  Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few
+	  errors in sample cdr config file.
+
+2007-08-20 21:57 +0000 [r80088]  Russell Bryant <russell at digium.com>
+
+	* apps/app_queue.c: Fix the build of app_queue
+
+2007-08-20 21:39 +0000 [r80049-80086]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: After a discussion on #asterisk-dev, it was
+	  decided that this should be in 1.4 as well. (issue #10424,
+	  reported and patched by irroot)
+
+	* apps/app_queue.c: Found a pointless ternary if. member->dynamic
+	  was set to 1 and has no opportunity to change between then and
+	  this line, so "dynamic" will ALWAYS be output.
+
+2007-08-20 16:08 +0000 [r80047]  Jason Parker <jparker at digium.com>
+
+	* configs/extensions.conf.sample: (issue #10499) Reported by:
+	  casper Patches: extensions.conf.sample.diff uploaded by casper
+	  (license 55) Update CLI examples in extensions.conf.sample to
+	  reflect command changes.
+
+2007-08-20 15:34 +0000 [r80044]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Ukrainian language voicemail support.
+	  (closes issue #10458, reported and patched by Oleh)
+
+2007-08-20 02:42 +0000 [r79998]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Missing curly braces. Oops. (Reported by
+	  snuffy via IRC)
+
+2007-08-18 14:30 +0000 [r79947]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Don't allocate vmu for messagecount when we
+	  could just use the stack instead (closes issue #10490) Also,
+	  remove a useless (and leaky) SQLAllocHandle (closes issue #10480)
+
+2007-08-17 21:01 +0000 [r79912]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c: Avoid a crash in the handling of DTMF based
+	  Caller ID. It is valid for ast_read to return NULL in the case
+	  that the channel has been hung up. (crash reported by
+	  anonymouz666 on IRC in #asterisk-dev)
+
+2007-08-17 19:14 +0000 [r79906]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Patch allows for more seamless transition
+	  from file storage voicemail to ODBC storage voicemail. If a
+	  retrieval of a greeting from the database fails, but the file is
+	  found on the file system, then we go ahead an insert the greeting
+	  into the database. The result of this is that people who switch
+	  from file storage to ODBC storage do not need to rerecord their
+	  voicemail greetings.
+
+2007-08-17 19:12 +0000 [r79902-79904]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c, main/utils.c, include/asterisk/strings.h:
+	  Don't send a semicolon over the wire in sip notify messages.
+	  Caused by fix for issue 9938. I basically took the code that
+	  existed before 9938 was fixed, and copied it into a new function
+	  - ast_unescape_semicolon There should be very few places this
+	  will be needed (pbx_config does NOT need this (see issue 9938 for
+	  details)) Issue 10430, patch by me, with help/ideas from murf
+	  (thanks murf).
+
+	* channels/chan_local.c: Re-add the setting of callerid name and
+	  number. Issue 10485, reported by and fix explained by paradise.
+
+2007-08-17 13:37 +0000 [r79857]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_sip.c: Fix some crashes in chan_sip. This patch
+	  changes various places that add items to the scheduler to ensure
+	  that they don't overwrite the ID of a previously scheduled item.
+	  If there is one, it should be removed. (closes issue #10391,
+	  closes issue #10256, probably others, patch by me)
+
+2007-08-17 08:22 +0000 [r79833]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c: sometimes we don't need to signal dtmf
+	  tones to asterisk, we just want them to go through as inband.
+	  Otherwise they might be generated by the other channel partner
+	  and then there is a double tone.
+
+2007-08-16 22:32 +0000 [r79756-79792]  Russell Bryant <russell at digium.com>
+
+	* res/res_musiconhold.c: Fix a little race condition that could
+	  cause a crash if two channels had MOH stopped at the same time
+	  that were using a class that had been marked for deletion when
+	  its use count hits zero.
+
+	* res/res_musiconhold.c: This patch fixes a bug where reloading the
+	  module with "module reload" did not delete classes from memory
+	  that were no longer in the config. This patch fixes that problem
+	  as well as another one. Previously, if you reloaded MOH using the
+	  "moh reload" CLI command, which behaved differently than "module
+	  reload ...", MOH had to be stopped on every channel and started
+	  again immediately. However, there was no way to tell what class
+	  was being used, so they would all fall back to the default class.
+	  (closes issue #10139) Reported by: blitzrage Patches:
+	  asterisk-10139-advanced.diff.txt uploaded by jamesgolovich
+	  (license 176) Tested by: jamesgolovich
+
+	* channels/chan_iax2.c: Fix more deadlocks in chan_iax2 that were
+	  introduced by making frame handling and scheduling
+	  multi-threaded. Unfortunately, we have to do some expensive
+	  deadlock avoidance when queueing frames on to the ast_channel
+	  owner of the IAX2 pvt struct. This was already handled for
+	  regular frames, but ast_queue_hangup and ast_queue_control were
+	  still used directly. Making these changes introduced even more
+	  places where the IAX2 pvt struct can disappear in the context of
+	  a function holding its lock due to calling a function that has to
+	  unlock/lock it to avoid deadlocks. I went through and fixed all
+	  of these places to account for this possibility. (issue #10362,
+	  patch by me)
+
+2007-08-16 21:16 +0000 [r79690-79748]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_agent.c: Fixes a problem where agents would get
+	  stuck busy due to their wrapuptime being longer than the queue's
+	  wrapuptime and ringinuse=no for the queue. (closes issue #10215,
+	  reported by Doug, repaired by me) Special thanks to fkasumovic
+	  for pointing out the source of the problem and to bweschke for
+	  helping to come up with a solution!
+
+	* apps/app_voicemail.c: base_encode is not trying to open a log
+	  file, so we should not call it a log file in the warning.
+	  (related to issue #10452, reported by bcnit)
+
+2007-08-16 09:37 +0000 [r79665]  Philippe Sultan <philippe.sultan at gmail.com>
+
+	* res/res_jabber.c: A fix for two critical problems detected while
+	  working with Daniel McKeehan in issue #10184. Upon priority
+	  change, the resource list is not NULL terminated when moving an
+	  item to the end of the list. This makes Asterisk endlessy loop
+	  whenever it needs to read the list. Jids with different resource
+	  and priority values, like in Gmail's and GoogleTalk's jabber
+	  clients put that problem in evidence. Upon reception of a 'from'
+	  attribute with an empty resource string, Asterisk crashes when
+	  trying to access the found->cap pointer if the resource list for
+	  the given buddy is not empty. This situation is perfectly valid
+	  and must be handled. The Gizmoproject's jabber client put that
+	  problem in evidence. Also added a few comments in the code as
+	  well as a handle for the capabilities from Gmail's jabber client,
+	  which are stored in a caps:c tag rather than the usual c tag.
+	  Closes issue #10184.
+
+2007-08-16 08:21 +0000 [r79642]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/misdn/ie.c: 0x80 + protocol is wrong for USERUSER when
+	  we want to send IA5 Chars.
+
+2007-08-15 14:40 +0000 [r79553]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c: (closes issue #10440) Reported by: irroot (closes
+	  issue #10454) Reported by: flo_turc Increase maximum timestamp
+	  skew to 120. 20 was apparently far too low.
+
+2007-08-15 14:26 +0000 [r79527]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Fixed an error in the Russian language
+	  voicemail intro. (issue #10458, reported and patched by Oleh)
+
+2007-08-15 14:18 +0000 [r79523]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: (closes issue #10456) Reported by: irroot
+	  Patches: sip_timeout.patch uploaded by irroot (license 52) Change
+	  hardcoded timer value to defined value. I'm doing this in 1.4 as
+	  well so if it needs to be changed in the future this place would
+	  not have been forgotten.
+
+2007-08-14 18:49 +0000 [r79436-79470]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix another spot where an iax2_peer would
+	  be leaked if realtime was in use.
+
+	* channels/chan_iax2.c: Fix some memory leaks throughout chan_iax2
+	  related to the use of realtime. I found these while working on
+	  iax2_peer object reference tracking.
+
+2007-08-14 15:27 +0000 [r79397]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: (closes issue #10415) Reported by: atis
+	  Revert fix for #10327 as it causes more issues then it solves.
+
+2007-08-13 22:40 +0000 [r79363]  Steve Murphy <murf at digium.com>
+
+	* pbx/pbx_ael.c: memset really, really needs to be used here.
+
+2007-08-13 21:57 +0000 [r79334]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_speech.c, apps/app_speech_utils.c,
+	  include/asterisk/speech.h: Instead of accepting a single DTMF
+	  character accept a full string.
+
+2007-08-13 20:37 +0000 [r79272-79301]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Don't call find_peer in
+	  registry_authrequest with the pvt lock held to avoid a deadlock.
+
+	* channels/chan_iax2.c: Release the pvt lock before calling
+	  find_peer in register_verify to avoid a deadlock. Also, remove
+	  some unnecessary locking in auth_fail that was only done
+	  recursively.
+
+	* channels/chan_iax2.c: Don't call find_peer within update_registry
+	  with a pvt lock held. This can cause a deadlock as the code will
+	  eventually call find_callno.
+
+	* channels/chan_iax2.c: I am fighting deadlocks in chan_iax2. I
+	  have tracked them down to a single core issue. You can not call
+	  find_callno() while holding a pvt lock as this function has to
+	  lock another (every) other pvt lock. Doing so can lead to a
+	  classic deadlock. So, I am tracking down all of the code paths
+	  where this can happen and fixing them. The fix I committed
+	  earlier today was along the same theme. This patch fixes some
+	  code down the path of authenticate_reply.
+
+2007-08-13 17:49 +0000 [r79255]  Steve Murphy <murf at digium.com>
+
+	* pbx/ael/ael-test/ref.ael-vtest21 (added),
+	  pbx/ael/ael-test/ref.ael-test19,
+	  pbx/ael/ael-test/ael-vtest21/extensions.ael (added),
+	  pbx/ael/ael-test/ael-vtest21 (added),
+	  pbx/ael/ael-test/ref.ael-vtest17,
+	  pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+	  pbx/ael/ael-test/ref.ael-test11, pbx/pbx_ael.c,
+	  pbx/ael/ael-test/ref.ael-test14, utils/ael_main.c: This patch
+	  fixes bug 10411. I added a new regression test, some regression
+	  test cleanups
+
+2007-08-13 15:28 +0000 [r79214]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix a potential deadlock in socket_process.
+	  check_provisioning can eventually call find_callno. You can't
+	  hold a pvt lock while calling find_callno because it goes through
+	  and locks every single one looking for a match.
+
+2007-08-13 14:51 +0000 [r79174-79207]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_speech.c, apps/app_speech_utils.c,
+	  include/asterisk/speech.h: Add an API call to allow the engine to
+	  know that DTMF was received.
+
+	* channels/chan_oss.c, channels/chan_mgcp.c, channels/chan_phone.c,
+	  channels/chan_local.c, channels/chan_misdn.c,
+	  channels/chan_zap.c, channels/chan_sip.c, channels/chan_skinny.c,
+	  channels/chan_h323.c, channels/chan_gtalk.c,
+	  channels/chan_iax2.c: (closes issue #10437) Reported by: haklin
+	  Don't set the callerid name and number a second time on a newly
+	  created channel. ast_channel_alloc itself already sets it and
+	  setting it twice would cause a memory leak.
+
+2007-08-11 05:23 +0000 [r79142]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* res/res_odbc.c: Ensure the connection gets marked as used at
+	  allocation time (closes issue #10429, report and fix by
+	  mnicholson)
+
+2007-08-10 20:53 +0000 [r79044-79099]  Steve Murphy <murf at digium.com>
+
+	* main/channel.c, pbx/pbx_spool.c, include/asterisk/channel.h: From
+	  a user complaint on #asterisk, I have forced pbx_spool to explain
+	  what reason codes mean, when they are logged
+
+	* main/cdr.c: Re bug behavior mentioned in #asterisk, made this
+	  tweak to code, to prevent hundreds of log messages from being
+	  generated
+
+	* main/cdr.c: This will help debug; from a question asked on
+	  #asterisk
+
+2007-08-10  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.10.1 released.
+	
+2007-08-10 15:20 +0000 [r78995]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/lock.h: The last set of changes that I made to
+	  "core show locks" made it not able to track mutexes unless they
+	  were declared using AST_MUTEX_DEFINE_STATIC. Locks initialized
+	  with ast_mutex_init() were not tracked. It should work now.
+
+2007-08-10 14:15 +0000 [r78951-78955]  Joshua Colp <jcolp at digium.com>
+
+	* main/file.c: Don't bother having the core pass through or emulate
+	  begin DTMF frames when in an ast_waitstream. It only cares about
+	  the end of DTMF.
+
+	* configs/queues.conf.sample: (closes issue #10422) Reported by:
+	  bhowell Add note to sample configuration about module load order
+	  and how it can cause perfectly good queue members to be marked as
+	  invalid.
+
+2007-08-10 13:24 +0000 [r78936]  Christian Richter <christian.richter at beronet.com>
+
+	* channels/chan_misdn.c, channels/misdn/ie.c,
+	  channels/misdn/isdn_msg_parser.c: fixed a bug with the useruser
+	  information element. We send them now also in the disconnect
+	  message.
+
+2007-08-09 23:47 +0000 [r78907]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Improved a bit of logic regarding
+	  comma-separated mailboxes in has_voicemail. Also added some
+	  braces to some compound if statements since unbraced if
+	  statements scare me in general.
+
+2007-08-09 23:10 +0000 [r78891]  Steve Murphy <murf at digium.com>
+
+	* Makefile: This fixes bug 10416; thanks to mvanbaak for the pretty
+	  output
+
+2007-08-09 22:03 +0000 [r78826-78860]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Removing some extra debug code I left in my
+	  last commit
+
+	* apps/app_voicemail.c: Quite a few changes regarding IMAP storage.
+	  1. instead of using inboxcount as the core message counting
+	  function, we use messagecount instead. This makes it possible to
+	  count messages in folders besides just INBOX and Old. 2.
+	  inboxcount and hasvoicemail now use messagecount as their means
+	  of determining return values. 3. Added a copy_message function
+	  for IMAP storage. Unfortunately I don't have the means to test
+	  it, but it seems like a pretty straightforward function. 4.
+	  Removed a #ifndef IMAP_STORAGE and matching #endif from
+	  leave_voicemail for a couple of reasons. One, we want to support
+	  copying mail to multiple IMAP boxes, and two, IMAP was broken
+	  because a STORE macro had been moved into this section of code.
+
+	* channels/chan_sip.c: I broke canreinvite...Now I'm fixing it. I
+	  put some new code in the wrong place and so I've reverted the
+	  canreinvite section to how it was and put my new code where it
+	  should be.
+
+2007-08-09 17:58 +0000 [r78717-78778]  Russell Bryant <russell at digium.com>
+
+	* apps/app_voicemail.c: add a comment to indicate that inboxcount
+	  for ODBC_STORAGE needs to be fixed to support multiple mailboxes
+
+	* apps/app_voicemail.c: Fix subscriptions to multiple mailboxes for
+	  ODBC_STORAGE. Also, leave a comment for this to be fixed for
+	  IMAP_STORAGE, as well. I left IMAP alone since I know MarkM was
+	  working on this code right now for another reason. This is broken
+	  even worse in trunk, but for a different reason. The fact that
+	  the mailbox option supported multiple mailboxes is completely not
+	  obvious from the code in the channel drivers. Anyway, I will fix
+	  that in another commit ...
+
+	* apps/app_meetme.c: Fix a problem with the combination of the 'F'
+	  option to pass DTMF through a conference and options that use
+	  DTMF to activate various features. The problem was that the BEGIN
+	  frame would be passed through, but the END frame would get
+	  intercepted to activate a feature. Then, the other conference
+	  members would hear DTMF for forever, which they didn't seem to
+	  like very much. (closes issue #10400, reported by stevefeinstein,
+	  fixed by me)
+
+2007-08-08 19:29 +0000 [r78646]  Jason Parker <jparker at digium.com>
+
+	* doc/jabber.txt: Fix mogs email address.
+
+2007-08-08 18:16 +0000 [r78575-78620]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: Fixed some compiler warnings so that
+	  compiling with dev-mode and IMAP storage would not have any
+	  errors. This section of code may get changed again shortly since
+	  my change uncovers a rather silly bit of logic.
+
+	* apps/app_queue.c: Changing a bit of logic so that someone will
+	  NEVER exit the queue on timeout unless they have enabled the 'n'
+	  option. This commit relates to issue #10320. Thanks to
+	  jfitzgibbon for detailing the idea behind this code change.
+
+2007-08-08 13:51 +0000 [r78569]  Joshua Colp <jcolp at digium.com>
+
+	* configs/sip.conf.sample: (closes issue #10335) Reported by:
+	  adamgundy Update sip.conf to include another scenario where
+	  directrtpsetup will fail.
+
+2007-08-07  Russell Bryant <russell at digium.com>
+
+	* Asterisk 1.4.10 released.
+	
+2007-08-07 20:57 +0000 [r78488]  Russell Bryant <russell at digium.com>
+
+	* res/res_config_odbc.c: Fix the build of this module on 64-bit
+	  platforms
+
+2007-08-07 19:43 +0000 [r78450]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_voicemail.c: The logic behind inboxcount's return value
+	  was reversed in has_voicemail and message_count. (closes issue
+	  #10401, reported by st1710, patched by me)
+
+2007-08-07 19:34 +0000 [r78437]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* res/res_odbc.c: Don't free the environment handle when the
+	  connection fails, because other connections might be depending
+	  upon it
+
+2007-08-07 19:11 +0000 [r78416]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_sip.c: Allow chan_sip to build in devmode
+
+2007-08-07 19:09 +0000 [r78415]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c, res/res_config_odbc.c,
+	  apps/app_directory.c: Reconnection doesn't happen automatically
+	  when a DB goes down (fixes issue #9389)
+
+2007-08-07 18:25 +0000 [r78375]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c: Properly check the capabilities count to
+	  avoid a segfault. (ASA-2007-019)
+
+2007-08-07 17:45 +0000 [r78371]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_zap.c, /: Merged revisions 78370 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r78370 | russell | 2007-08-07 12:44:04 -0500 (Tue, 07 Aug 2007) |
+	  4 lines Revert patch committed for issue #9660. It broke E&M
+	  trunks. (closes issue #10360) (closes issue #10364) ........
+
+2007-08-06 21:41 +0000 [r78275]  Joshua Colp <jcolp at digium.com>
+
+	* main/channel.c: Add additional DTMF log messages to help when
+	  debugging issues.
+
+2007-08-06 20:44 +0000 [r78184-78242]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix an issue where dynamic threads can get
+	  free'd, but still exist in the dynamic thread list. (closes issue
+	  #10392, patch from Mihai, with credit to his colleague, Pete)
+
+	* include/asterisk/linkedlists.h: Fix the return value of
+	  AST_LIST_REMOVE(). This shouldn't be causing any problems,
+	  though, because the only code that uses the return value only
+	  checks to see if it is NULL. (closes issue #10390, pointed out by
+	  mihai)
+
+2007-08-06 16:32 +0000 [r78182]  Joshua Colp <jcolp at digium.com>
+
+	* channels/chan_sip.c: It is possible for a transfer to occur
+	  before the remote device has our tag in which case they send none
+	  in the transfer. In this case we need to not fail the transfer
+	  dialog lookup.
+
+2007-08-06 16:30 +0000 [r78180]  Jason Parker <jparker at digium.com>
+
+	* main/config.c: Fix an issue with using UpdateConfig (manager
+	  action) where escaped semicolons in a config would be converted
+	  to just semicolons (\; to ;) Issue 9938
+
+2007-08-06 15:27 +0000 [r78166-78172]  Joshua Colp <jcolp at digium.com>
+
+	* main/rtp.c: (closes issue #10355) Reported by: wdecarne Now that
+	  we pass through RTP timestamp information we need to make the
+	  allowed timestamp skew considerably less. There are situations
+	  where a source may change and due to the timestamp difference the
+	  receiver will experience an audio gap since we did not indicate
+	  by setting the marker bit that the source changed.
+
+	* configure, configure.ac: (closes issue #10383) Reported by: rizzo
+	  Include stdlib.h so NULL gets defined for gethostbyname_r checks.
+
+2007-08-06 13:33 +0000 [r78164]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: Fixed a mistake I made in realtime_peer
+	  which caused it to return NULL every time. Thanks to Jon Fealy
+	  for emailing me the correction.
+
+2007-08-05 14:18 +0000 [r78146]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* cdr/cdr_pgsql.c: Portability fix for devmode compiling (closes
+	  bug #10382)
+
+2007-08-05 04:15 +0000 [r78143]  Russell Bryant <russell at digium.com>
+
+	* include/asterisk/lock.h: Fix compilation failure when
+	  MALLOC_DEBUG is enabled, but DEBUG_THREADS is not
+
+2007-08-05 03:29 +0000 [r78139]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* channels/chan_sip.c: If peer is not found, the error message is
+	  misleading (should be peer not found, not ACL failure)
+
+2007-08-03 20:25 +0000 [r78103]  Mark Michelson <mmichelson at digium.com>
+
+	* main/config.c, channels/chan_sip.c, include/asterisk/config.h:
+	  Changed the behavior of sip's realtime_peer function to match the
+	  corresponding way of matching for non-realtime peers. Now matches
+	  are made on both the IP address and port number, or if the
+	  insecure setting is set to "port" then just match on the IP
+	  address. In order to accomplish this, I also added a new API
+	  call, ast_category_root, which returns the first variable of an
+	  ast_category struct
+
+2007-08-03 20:14 +0000 [r78028-78101]  Russell Bryant <russell at digium.com>
+
+	* apps/app_voicemail.c: (closes issue #10194) Reported by:
+	  blitzrage Patches: bug0010194 uploaded by vovochka Tested by:
+	  blitzrage Fix a problem when you call Voicemail() with multiple
+	  mailboxes specified and ODBC_STORAGE is in use. The audio part of
+	  the message was only given to the first mailbox specified.
+
+	* main/utils.c, include/asterisk/lock.h, main/astmm.c: Add some
+	  improvements to lock debugging. These changes take effect with
+	  DEBUG_THREADS enabled and provide the following: * This will keep
+	  track of which locks are held by which thread as well as which
+	  lock a thread is waiting for in a thread-local data structure. A
+	  reference to this structure is available on the stack in the
+	  dummy_start() function, which is the common entry point for all
+	  threads. This information can be easily retrieved using gdb if
+	  you switch to the dummy_start() stack frame of any thread and
+	  print the contents of the lock_info variable. * All of the
+	  thread-local structures for keeping track of this lock
+	  information are also stored in a list so that the information can
+	  be dumped to the CLI using the "core show locks" CLI command.
+	  This introduces a little bit of a performance hit as it requires
+	  additional underlying locking operations inside of every
+	  lock/unlock on an ast_mutex. However, the benefits of having this
+	  information available at the CLI is huge, especially considering
+	  this is only done in DEBUG_THREADS mode. It means that in most
+	  cases where we debug deadlocks, we no longer have to request
+	  access to the machine to analyze the contents of ast_mutex_t
+	  structures. We can now just ask them to get the output of "core
+	  show locks", which gives us all of the information we needed in
+	  most cases. I also had to make some additional changes to astmm.c
+	  to make this work when both MALLOC_DEBUG and DEBUG_THREADS are
+	  enabled. I disabled tracking of one of the locks in astmm.c
+	  because it gets used inside the replacement memory allocation
+	  routines, and the lock tracking code allocates memory. This
+	  caused infinite recursion.
+
+	* channels/chan_iax2.c: Only pass through HOLD and UNHOLD control
+	  frames when the mohinterpret option is set to "passthrough". This
+	  was pointed out by Kevin in the middle of a training session.
+
+	* channels/chan_iax2.c: Don't reuse the timespec that was set to 0
+	  in the previous timedwait as it will just return immediately.
+	  Also, fix some logic so the thread's lock isn't unlocked twice in
+	  the weird case of dynamic threads getting acquired right after a
+	  timeout. (pointed out by SteveK)
+
+2007-08-02 21:53 +0000 [r77993-77996]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c, configs/skinny.conf.sample: Make sure we
+	  actually allow 6 chars to be sent. Also make note of the "A"
+	  option of date format. Issue 9779, modifications by DEA, wedhorn,
+	  and myself.
+
+	* channels/chan_skinny.c: If a device disconnects, the session will
+	  go away. If this happens during call setup, we need to give up.
+	  Issue 10325.
+
+2007-08-02 19:25 +0000 [r77949]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix the case where a dynamic thread times
+	  out waiting for something to do during the first time it runs.
+	  This shouldn't ever happen, but we should account for it anyway.
+	  (pointed out by pete, who works with mihai)
+
+2007-08-02 18:42 +0000 [r77947]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c: Make sure we clear the prompt status
+	  message on a hangup. Also rearrange messages to better fit with
+	  what a wireshark trace shows it should be. Issue 10299, initial
+	  patch and solution by sbisker, modified by me to fit with
+	  wireshark trace.
+
+2007-08-02 18:21 +0000 [r77945]  Steve Murphy <murf at digium.com>
+
+	* main/fskmodem.c, /: Merged revisions 77942 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1
+	  line This patch hopefully solves 10141; The user is running with
+	  it, and it doesn't appear to harm asterisk's operation, and may
+	  prevent a crash. I'll store it in 1.2, as we have shut down
+	  support on 1.2, but since I developed the patch before support
+	  finished, and it might affect 1.4 and trunk, I'm going ahead with
+	  it. ........
+
+2007-08-02 18:04 +0000 [r77939-77943]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix another race condition in the handling
+	  of dynamic threads. If the dynamic thread timed out waiting for
+	  something to do, but was acquired to perform an action
+	  immediately afterwords, then wait on the condition again to give
+	  the other thread a chance to finish setting up the data for what
+	  action this thread should perform. Otherwise, if it immediately
+	  continues, it will perform the wrong action. (reported on IRC by
+	  mihai, patch by me) (related to issue #10289)
+
+	* channels/chan_iax2.c: Add another sanity check to
+	  vnak_retransmit(). This check ensures that frames that have
+	  already been marked for deletion don't get retransmitted. (closes
+	  issue #10361, patch from mihai)
+
+2007-08-02 15:15 +0000 [r77890-77894]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c: Make sure that we show the correct
+	  extension if dialed from a macro "From: 5555" rather than "From:
+	  s" Issue 10358, initial patch by DEA, reworked by me to use S_OR,
+	  tested by sbisker
+
+	* channels/chan_skinny.c: Put in some additional debug information
+	  for softkey/stimulus messages. Issue 10291, patch by DEA.
+
+2007-08-01 22:16 +0000 [r77887]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix some race conditions which have been
+	  causing weird problems in chan_iax2. The most notable problem is
+	  that people have been seeing storms of VNAK frames being sent due
+	  to really old frames mysteriously being in the retransmission
+	  queue and never getting removed. It was possible that a dynamic
+	  thread got created, but did not acquire its lock before the
+	  thread that created it signals it to perform an action. When this
+	  happens, the thread will sleep until it hits a timeout, and then
+	  get destroyed. So, the action never gets performed and in some
+	  cases, means a frame doesn't get transmitted and never gets freed
+	  since the scheduler never gets a chance to reschedule
+	  transmission. Another less severe race condition is in the
+	  handling of a timeout for a dynamic thread. It was possible for
+	  it to be acquired to perform at action at the same time that it
+	  hit a timeout. When this occurs, whatever action it was acquired
+	  for would never get performed. (patch contributed by Mihai and
+	  SteveK) (closes issue #10289) (closes issue #10248) (closes issue
+	  #10232) (possibly related to issue #10359)
+
+2007-08-01 22:14 +0000 [r77886]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* apps/app_voicemail.c: Voicemail with ODBC_STORAGE defined does
+	  not compile cleanly (missing def)
+
+2007-08-01 21:08 +0000 [r77883]  Jason Parker <jparker at digium.com>
+
+	* channels/chan_skinny.c: Fix an issue that caused one-way audio on
+	  some newer devices (specifically the 7921), due to sending
+	  packets in the wrong order during hangup. Also make sure we clear
+	  tones/messages on the correct line/instance. Issue 10291, patch
+	  by DEA, tested by sbisker and myself.
+
+2007-08-01 18:08 +0000 [r77863-77871]  Joshua Colp <jcolp at digium.com>
+
+	* main/cli.c: (closes issue #10351) Reported by: ftarz Some
+	  platforms don't like it when you pass NULL to vsnprintf so pass
+	  "" instead.
+
+	* include/asterisk/threadstorage.h, channels/chan_mgcp.c,
+	  apps/app_voicemail.c, main/acl.c, utils/smsq.c,
+	  channels/chan_iax2.c: Add some fixes for building on Solaris.
+
+	* main/utils.c: Whoops, I meant R_5 not R5.
+
+	* configure, configure.ac: And for my last trick... make sure that
+	  if gethostbyname_r is exported by a library that it is used.
+
+	* configure, include/asterisk/autoconfig.h.in, configure.ac,
+	  main/utils.c: Extend autoconf logic to determine which version of
+	  gethostbyname_r is on the system.
+
+2007-08-01 14:08 +0000 [r77852-77854]  Mark Michelson <mmichelson at digium.com>
+
+	* apps/app_queue.c: Fixes an issue I introduced to queues wherein a
+	  queue with joinempty=yes would kick people out of the queue
+	  because of erroneously thinking the 'n' option was in use.
+	  (closes issue #10320, reported by jfitzgibbon, patched by me,
+	  tested by blitzrage and me) Thank you blitzrage for all the
+	  testing you've done lately with queues! It's much appreciated!
+
+	* apps/app_queue.c: If a queue uses dynamic realtime members, then
+	  the member list should be updated after each attempt to call the
+	  queue. This fixes an issue where if a caller calls into a queue
+	  where no one is logged in, they would wait forever even if a
+	  member logged in at some point. (closes issue #10346, reported by
+	  and tested by blitzrage, patched by me)
+
+2007-07-31 21:09 +0000 [r77845-77846]  Jim Dixon <telesistant at hotmail.com>
+
+	* apps/app_rpt.c: Much newer version, 0.70 with much additions
+
+	* main/dsp.c, channels/chan_zap.c: Made VAST improvements in DTMF
+	  receiver in RADIO_RELAX mode (thanx Steve W9SH), and oversight in
+	  logic in TONE_VERIFY/RELAX mode in chan_zap.
+
+2007-07-31 20:59 +0000 [r77844]  Steve Murphy <murf at digium.com>
+
+	* /, contrib/scripts/ast_grab_core: Merged revisions 77842 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r77842 | murf | 2007-07-31 13:19:35 -0600 (Tue, 31 Jul 2007) | 1
+	  line This probably isn't super-general, but it's a first stab at
+	  using kill -11 to generate a core file instead of gcore. ........
+
+2007-07-31 16:17 +0000 [r77831]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_speech.c, include/asterisk/speech.h: Add a flag to the
+	  speech API that allows an engine to set whether it received
+	  results or not.
+
+2007-07-31 15:53 +0000 [r77827]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* build_tools/cflags.xml: DETECT_DEADLOCKS can't be enabled without
+	  DEBUG_THREADS or it does nothing
+
+2007-07-31 15:21 +0000 [r77824]  Mark Michelson <mmichelson at digium.com>
+
+	* channels/chan_sip.c: This patch makes Asterisk send 100 Trying
+	  provisional responses upon receipt of re-invites. This makes it
+	  so that if there are two or more Asterisk servers between
+	  endpoints, the Asterisk servers will not keep retransmitting the
+	  re-invites. (closes issue #10274, reported by cstadlmann, patched
+	  by me with approval from file)
+
+2007-07-30 20:17 +0000 [r77795]  Jason Parker <jparker at digium.com>
+
+	* main/say.c: Applications like SayAlpha() should not hang up the
+	  channel if you request an "unknown" character such as a comma.
+	  Instead, skip the character and move on. Issue 10083, initial
+	  patch by jsmith, modified by me.
+
+2007-07-30 20:16 +0000 [r77785-77794]  Russell Bryant <russell at digium.com>
+
+	* channels/chan_iax2.c: Fix an issue that could potentially cause
+	  corruption of the global iax frame queue. In the network_thread()
+	  loop, it traverses the list using the AST_LIST_TRAVERSE_SAFE
+	  macro. However, to remove an element of the list within this
+	  loop, it used AST_LIST_REMOVE, instead of
+	  AST_LIST_REMOVE_CURRENT, which I believe could leave some of the
+	  internal variables of the SAFE macro invalid. Mihai says that he
+	  already made this change in his local copy and it didn't help his
+	  VNAK storm issues, but I still think it's wrong. :)
+
+	* res/res_agi.c: (closes issue #10279) Reported by: seanbright
+	  Patches: res_agi.carefulwrite.1.4.07252007.patch uploaded by
+	  seanbright (license 71) res_agi.carefulwrite.trunk.07252007.patch
+	  uploaded by seanbright (license 71) Allow the "agi_network: yes"
+	  line to be printed out in the AGI debug output. Also, allow
+	  partial writes to be handled when writing out this line just like
+	  it is for all of the others.
+
+	* main/channel.c: file and I both committed changes for issue
+	  #10301. Remove a duplicated assignment to restore the original
+	  value of the previous channel.
+
+2007-07-30 18:43 +0000 [r77783]  Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+	* /, res/res_agi.c: Merged revisions 77782 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+	  r77782 | tilghman | 2007-07-30 13:40:54 -0500 (Mon, 30 Jul 2007)
+	  | 2 lines Revert change in revision 71656, even though it fixed a
+	  bug, because many people were depending upon the (broken)
+	  behavior. ........
+
+2007-07-30 17:29 +0000 [r77780]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c: (closes issue #10301) Reported by: fnordian
+	  Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
+	  (license 110) Additional changes by me Fix some problems in
+	  channel_find_locked() which can cause an infinite loop. The
+	  reference to the previous channel is set to NULL in some cases.
+	  These changes ensure that the reference to the previous channel
+	  gets restored before needing it again. I'm not convinced that the
+	  code that is setting it to NULL is really the right thing to do.
+	  However, I am making these changes to fix the obvious problem and
+	  just leaving an XXX comment that it needs a better explanation
+	  that what is there now.
+
+2007-07-30 17:11 +0000 [r77768-77778]  Joshua Colp <jcolp at digium.com>
+
+	* res/res_features.c: (closes issue #10327) Reported by: kkiely
+	  Instead of directly mucking with the extension/context/priority

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