[asterisk-commits] russell: tag 1.4.10.1 r78958 - in /tags/1.4.10.1: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Aug 10 09:22:13 CDT 2007
Author: russell
Date: Fri Aug 10 09:22:12 2007
New Revision: 78958
URL: http://svn.digium.com/view/asterisk?view=rev&rev=78958
Log:
importing files for 1.4.10.1 release
Added:
tags/1.4.10.1/.lastclean (with props)
tags/1.4.10.1/.version (with props)
tags/1.4.10.1/ChangeLog (with props)
Added: tags/1.4.10.1/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.10.1/.lastclean?view=auto&rev=78958
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--- tags/1.4.10.1/ChangeLog (added)
+++ tags/1.4.10.1/ChangeLog Fri Aug 10 09:22:12 2007
@@ -1,0 +1,10534 @@
+2007-08-10 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.10.1 released.
+
+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
+ of the channel we are transferring when it has a PBX simply call
+ ast_async_goto on it. This will ensure that the channel gets
+ handled properly and sent to the right place.
+
+ * main/channel.c: (closes issue #10301) Reported by: fnordian
+ Patches: asterisk-1.4.9-channel.c.patch uploaded by fnordian
+ (license 110) Restore previous behavior where if we failed to
+ lock the channel we wanted we would return to exactly the same
+ point as if we had just reentered the function.
+
+ * /, apps/app_macro.c: Merged revisions 77767 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r77767 | file | 2007-07-30 11:50:02 -0300 (Mon, 30 Jul 2007) | 4
+ lines (closes issue #10334) Reported by: ramonpeek Pass through
+ the return value from macro_exec through the MacroIf application.
+ ........
+
+2007-07-27 18:15 +0000 [r77571] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * res/res_odbc.c: Missing newline
+
+2007-07-27 17:04 +0000 [r77536-77540] Joshua Colp <jcolp at digium.com>
+
+ * cdr/cdr_pgsql.c: (closes issue #10310) Reported by: prashant_jois
+ Patches: cdr_pgsql.patch uploaded by prashant (license 114)
+ Finish the Postgresql connection after the log messages are
+ printed so we don't access invalid memory.
+
+ * channels/chan_sip.c: (closes issue #10323) Reported by: julianjm
+ Patches: chan_sip_device_state_hold_fix.v1.diff.txt uploaded by
+ julianjm (license 99) Clear ONHOLD flag when decrementing the
+ onHold peer count. If we did not do this the count may keep
+ decreasing.
+
+2007-07-27 14:30 +0000 [r77490] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: "re-invite" was misspelled
+
+2007-07-26 23:19 +0000 [r77460] Joshua Colp <jcolp at digium.com>
+
+ * main/channel.c: (closes issue #10302) Reported by: litnialex If a
+ DTMF end frame comes from a channel without a begin and it is
+ going to a technology that only accepts end frames (aka INFO)
+ then use the minimum DTMF duration if one is not in the frame
+ already.
+
+2007-07-26 22:16 +0000 [r77424-77429] Kevin P. Fleming <kpfleming at digium.com>
+
+ * doc/mp3.txt: change protocol for downloads as well
+
+ * doc/mp3.txt, sounds/Makefile: use new canonical name for download
+ server
+
+2007-07-26 21:23 +0000 [r77410] Russell Bryant <russell at digium.com>
+
+ * Makefile, build_tools/make_buildopts_h: AST_DEVMODE was defined
+ in trunk, but not in 1.4. When Asterisk is compiled under dev
+ mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
+ define it in the same way that trunk does. Also, revert the
+ change that added this define in the Makefile The advantage to
+ doing it this way is that buildopts.h gets installed when you
+ install Asterisk. Then, when building any out of tree modules, or
+ building asterisk-addons, these modules know which options the
+ rest of Asterisk was built with.
+
+2007-07-26 20:35 +0000 [r77380] Mark Michelson <mmichelson at digium.com>
+
+ * Makefile, main/logger.c: Fixes to get ast_backtrace working
+ properly. The AST_DEVMODE macro was never defined so the majority
+ of ast_backtrace never attempted compilation. The makefile now
+ defines AST_DEVMODE if configure was run with --enable-dev-mode.
+ Also, changes were made to acccomodate 64 bit systems in
+ ast_backtrace. Thanks to qwell, kpfleming, and Corydon76 for
+ their roles in allowing me to get this committed
+
+2007-07-26 19:32 +0000 [r77348-77350] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * main/logger.c: Missed one
+
+ * main/logger.c: Oops, that builtin define should be all-lowercase.
+
+2007-07-26 18:30 +0000 [r77318] Mark Michelson <mmichelson at digium.com>
+
+ * cdr/cdr_pgsql.c: Two consecutive calls to PQfinish could occur,
+ meaning free gets called on the same variable twice. This patch
+ sets the connection to NULL after calls to PQfinish so that the
+ problem does not occur. Also in this patch, prashant_jois
+ informed me that it is safe to pass a null pointer to PQfinish,
+ so I have removed the check for conn's existence from
+ my_unload_module. (closes issue 10295, reported by junky, patched
+ by me with input from prashant_jois)
+
+2007-07-25 22:39 +0000 [r77191] Steve Murphy <murf at digium.com>
+
+ * apps/app_meetme.c: This fix solves problem with intense squelch
+ noise when someone joins conf in bug 9430; We repro'd the problem
+ with meetme opts of 'CciMo'; Josh Colp supplied this patch, and
+ I'm applying it. It looks like playing the recorded username will
+ louse up the next thing played into the channel. Josh rearranged
+ the code so as to start things over before playing data directly
+ into the conference.
+
+2007-07-25 22:16 +0000 [r77176] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_speech_utils.c: (closes issue #10303) Reported by: jtodd
+ Add SPEECH_DTMF_TERMINATOR variable so the user can specify the
+ digit to terminate a DTMF string with. If none is specified then
+ no terminator will be used.
+
+2007-07-25 21:52 +0000 [r77154] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: chan->emulate_dtmf_duration is an unsigned int,
+ not a signed int, so use %u instead of %d in the format string
+
+2007-07-25 20:23 +0000 [r77116-77136] Jason Parker <jparker at digium.com>
+
+ * /: so are my fingers...
+
+ * /: autotagexternals script is still obviously misbehaving...
+
+ * /: use autotagged externals
+
+2007-07-25 17:14 +0000 [r77071] Joshua Colp <jcolp at digium.com>
+
+ * configure, acinclude.m4: Fix autoconf logic for finding OpenH323
+ when it is not in the first place searched (/usr/share/openh323).
+
+2007-07-25 09:34 +0000 [r77022] Luigi Rizzo <rizzo at icir.org>
+
+ * main/rtp.c: set the sequence number in a frame for all frame
+ types
+
+2007-07-25 00:18 +0000 [r76983] Steve Murphy <murf at digium.com>
+
+ * channels/chan_zap.c, /: Merged revisions 76978 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1
+ line this fixes bug 10293, where the error message because
+ defaultzone or loadzone was not defined was confusing ........
+
+2007-07-24 22:12 +0000 [r76891-76937] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /, include/asterisk/lock.h: Merged revisions 76934 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24
+ Jul 2007) | 2 lines Oops, res contains the error code, not errno.
+ I was wondering why a mutex was reporting "No such file or
+ directory"... ........
+
+ * main/app.c: Found another place where we should be using the
+ umask (thanks jcmoore)
+
+2007-07-24 Jason Parker <jparker at digium.com>
+
+ * Asterisk 1.4.9 released.
+
+2007-07-24 16:42 +0000 [r76803-76805] Jason Parker <jparker at digium.com>
+
+ * /: Blocked revisions 76802 via svnmerge ........ r76802 | qwell |
+ 2007-07-24 11:32:04 -0500 (Tue, 24 Jul 2007) | 3 lines Don't
+ create the Asterisk channel until we are starting the PBX on it.
+ (ASA-2007-018) ........
+
+ * channels/chan_iax2.c: Don't create the Asterisk channel until we
+ are starting the PBX on it. (ASA-2007-018)
+
+2007-07-24 16:26 +0000 [r76801] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Added a membercount variable to call_queue
+ struct which keeps track of the number of logged in members in a
+ particular queue. This makes it so that the 'n' option for
+ Queue() can act properly depending on which strategy is used. If
+ the strategy is roundrobin, rrmemory, or ringall, we want to ring
+ each phone once before moving on in the dialplan. However, if any
+ other strategy is used, we will only ring one phone since it
+ cannot be guaranteed that a different phone will ring on
+ subsequent attempts to ring a phone. As a side effect of this,
+ the QUEUE_MEMBER_COUNT dialplan function now just reads the
+ membercount variable instead of traversing through the member
+ list to figure out how many members there are. Special thanks to
+ blitzrage for helping to test this out. (closes issue #10127,
+ reported by bcnit, patched by me, tested by blitzrage)
+
+2007-07-23 22:38 +0000 [r76708] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * apps/app_voicemail.c: It was our stated intention for 1.4 that
+ files created in app_voicemail should depend upon the umask.
+ Unfortunately, mkstemp() creates files with mode 0600, regardless
+ of the umask. This corrects that deficiency.
+
+2007-07-23 18:59 +0000 [r76656] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Fix some incorrect softkey labels in
+ messages. Don't try to play dialtone in some unimplemented
+ features.
+
+2007-07-23 18:29 +0000 [r76654] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_agent.c: Merged revisions 76653 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul
+ 2007) | 4 lines (closes issue #5866) Reported by: tyler Do not
+ force channel format changes when a generator is present. The
+ generator may have changed the formats itself and changing them
+ back would cause issues. ........
+
+2007-07-23 17:57 +0000 [r76620] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Don't try to queue up hold/unhold frames
+ on a non-existent channel. Issue 10276.
+
+2007-07-23 17:48 +0000 [r76519-76618] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_morsecode.c: Allow app_morsecode to build on PPC Linux
+ by putting the value of the digit char in an int.
+
+ * /, channels/chan_sip.c: Merged revisions 76560 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6
+ lines (closes issue #10236) Reported by: homesick Patches:
+ rpid_1.4_75840.patch uploaded by homesick (license 91) Accept
+ Remote Party ID on guest calls. ........
+
+ * channels/chan_skinny.c: (closes issue #10268) Reported by:
+ mvanbaak Patches: chan_skinny_openbsd.diff uploaded by mvanbaak
+ (license 7) Add another OS that has to use the Macros for byte
+ ordering.
+
+2007-07-23 12:25 +0000 [r76485] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Use a signed integer for storing the number
+ of bytes in the packet read from the network. Using an unsigned
+ value here made it impossible to handle an error returned from
+ recvfrom(). Furthermore, in the case that recvfrom() did return
+ an error, this would cause a crash due to a heap overflow.
+ (closes issue #10265, reported by and fix suggested by
+ timrobbins)
+
+2007-07-22 21:42 +0000 [r76410] Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
+
+ * /: Blocked revisions 76409 via svnmerge ........ r76409 |
+ tilghman | 2007-07-22 16:39:55 -0500 (Sun, 22 Jul 2007) | 2 lines
+ We should not use C++ reserved words in API headers (closes issue
+ #10266) ........
+
+2007-07-21 02:02 +0000 [r76227] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 76226 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r76226 | russell | 2007-07-20 21:01:46 -0500 (Fri, 20 Jul 2007) |
+ 4 lines Backport a fix for a memory leak that was fixed in trunk
+ in reivision 76221 by rizzo. The memory used for the localaddr
+ list was not freed during a configuration reload. ........
+
+2007-07-20 21:36 +0000 [r76211] Steve Murphy <murf at digium.com>
+
+ * sounds/Makefile: This patch from 10249 is worth applying! It
+ prevents downloading sound files if they are already downloaded.
+ Darn Practical, if you ask me
+
+2007-07-20 21:03 +0000 [r76174-76178] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Allow getting a call from an existing
+ "sub" channel. Cancel ringing if endpoint hangs up before
+ answering. Fixes were backported from trunk (there was apparently
+ a bit of confusion during merge of a previous patch). (closes
+ issue #10241)
+
+ * main/manager.c: Eliminate a compiler warning with gcc 4.2 by
+ constifying a char *
+
+ * channels/chan_skinny.c: It's possible for sub->owner to be NULL
+ here if you cancel the call immediately after/during sending a
+ digit.
+
+2007-07-20 18:42 +0000 [r76139] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_directory.c: When using users.conf for the entries in
+ the directory, if multiple users had the same last name, only the
+ first user listed would be available in the directory. (closes
+ issue #10200, reported by mrskippy, patched by me)
+
+2007-07-20 18:22 +0000 [r76132] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Use the define that specifies the default length
+ of an artificially created DTMF digit in the ast_senddigit()
+ function. The define is set to 100ms by default, which is the
+ same thing that this function was using. But, using the define
+ lets changes take effect in this case, as well as the others
+ where it was already used.
+
+2007-07-20 17:20 +0000 [r76054-76087] Joshua Colp <jcolp at digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 76080 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r76080 | file | 2007-07-20 14:16:48 -0300 (Fri, 20 Jul 2007) | 6
+ lines (closes issue #10247) Reported by: fkasumovic Patches:
+ chan_sip.patch uploaded by fkasumovic (license #101) Drop any
+ peer realm authentication entries when reloading so multiple
+ entries do not get added to the peer. ........
+
+ * res/res_convert.c: (closes issue #10246) Reported by: fkasumovic
+ Patches: res_conver.patch uploaded by fkasumovic (license #101)
+ Use the last occurance of . to find the extension, not the first
+ occurance.
+
+ * apps/app_queue.c: Move makeannouncement variable declaration to
+ proper place.
+
+2007-07-19 20:36 +0000 [r75980] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Remove some duplicate code.
+
+2007-07-19 18:59 +0000 [r75969-75978] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: The diff on this looks pretty big but all I did
+ was remove a pointless if statement (always evaluates true).
+
+ * apps/app_queue.c: Changes in handling return values of several
[... 9733 lines stripped ...]
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