[asterisk-commits] russell: tag 1.4.17 r95953 - in /tags/1.4.17: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Jan 2 14:30:24 CST 2008
Author: russell
Date: Wed Jan 2 14:30:24 2008
New Revision: 95953
URL: http://svn.digium.com/view/asterisk?view=rev&rev=95953
Log:
Importing files for 1.4.17 release
Added:
tags/1.4.17/.lastclean (with props)
tags/1.4.17/.version (with props)
tags/1.4.17/ChangeLog (with props)
Added: tags/1.4.17/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.17/.lastclean?view=auto&rev=95953
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--- tags/1.4.17/ChangeLog (added)
+++ tags/1.4.17/ChangeLog Wed Jan 2 14:30:24 2008
@@ -1,0 +1,14559 @@
+2008-01-02 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.17 released.
+
+2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Allocate a SIP refer structure when
+ performing a transfer using BYE with Also so that the transfer
+ information is properly stored. (AST-2008-028) (closes issue
+ #11637) Reported by: greyvoip
+
+2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: A change to improve the accuracy of queue
+ logging in the case where a member does not answer during the
+ specified timeout period. Prior to this change, there was a small
+ chance that the member name recorded in this case would be blank.
+ Also prior to this change, if using the ringall strategy, if no
+ one answered the call during the specified timeout, the member
+ name listed in the queue log would randomly be one of the members
+ that was rung. (closes issue #11498, reported and tested by
+ hloubser, patched by me)
+
+2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson at digium.com>
+
+ * main/pbx.c: Avoiding a potentially bad locking situation.
+ ast_merge_contexts_and_delete writelocks the conlock, then calls
+ ast_hint_extension, which attempts to readlock the same lock.
+ Recursion with read-write locks is dangerous, so the inner lock
+ needs to be removed. I did this by copying the "guts" of
+ ast_hint_extension into ast_merge_contexts_and_delete (sans the
+ extra lock). (this change is inspired by the locking problems
+ seen in issue #11080, but I have no idea if this is the
+ problematic area experienced by the reporters of that issue)
+
+2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_env.c: Allow the default "0" to be returned if the
+ STAT fails (Closes issue #11659)
+
+2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Remove duplicate increment of the header
+ count in the add_header() function. (closes issue #11648)
+ Reported by: makoto Patch provided by sergee, committed patch by
+ me, inspired by comments from putnopvut
+
+2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: I found a bug while browsing the queue code and
+ managed to reproduce it in a small setup. If a queue uses the
+ ringall strategy, it was possible through unfortunate coincidence
+ for a single member at a given penalty level to make app_queue
+ think that all members at that penalty level were unavailable and
+ cause the members at the next penalty level to be rung. With this
+ patch, we will only move to the next penalty level if ALL the
+ members at a given penalty level are unreachable.
+
+2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell at digium.com>
+
+ * main/channel.c: Don't report a syntax error when an empty string
+ is passed to ast_get_group. Just return 0. (closes issue #11540)
+ Reported by: tzafrir Patches: group_empty.diff uploaded by
+ tzafrir (license 46) -- slightly changed by me
+
+2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson at digium.com>
+
+ * main/io.c: Fixing a typo in a comment.
+
+2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_h323.c: Include types.h in chan_h323 as without it
+ it can not be compiled on some operating systems like FreeBSD to
+ name one. (closes issue #11585) Reported by: sobomax Patches:
+ chan_h323.c.diff uploaded by sobomax (license 359)
+
+ * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
+ set or not on the dialog. It is possible for it to be a pointer
+ to NULL. (closes issue #11557) Reported by: FuriousGeorge
+
+2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c: Now that the contexts lock is a read/write lock, it
+ should not be locked here in ast_hint_state_changed(). This makes
+ it get locked recursively which now causes a deadlock. (closes
+ issue #11080, thanks to callguy for the access to a deadlocked
+ machine)
+
+ * include/asterisk/translate.h, main/translate.c: Use the constant
+ that I really meant to use here ...
+
+ * main/translate.c: Change ast_translator_best_choice() to only pay
+ attention to audio formats. This fixes a problem where Asterisk
+ claims that a translation path can not be found for channels
+ involving video. (closes issue #11638) Reported by: cwhuang
+ Tested by: cwhuang Patch suggested by cwhuang, with some
+ additional changes by me.
+
+2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/manager.c: make this comment explain the situation in an
+ even more explicit fashion
+
+2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher at digium.com>
+
+ * main/manager.c: Workaround for what is probably a glibc bug (but
+ we'll see this crop up again and again, if we don't add the
+ workaround). Reported by: rolek Patch by: tilghman (Closes issue
+ #11601, closes issue #11426)
+
+2007-12-26 19:04 +0000 [r94789-94801] Russell Bryant <russell at digium.com>
+
+ * main/autoservice.c: Just in case the AST_FLAG_END_DTMF_ONLY flag
+ was already set before starting autoservice, remember it and
+ ensure that the channel has the same setting when autoservice
+ gets stopped. (pointed out by d1mas, patched up by me)
+
+ * main/autoservice.c: When a channel is in autoservice, mark a flag
+ on the channel that says that we only care about the END of a
+ digit. That way, no magic digit emulation stuff will happen when
+ all we're doing is queueing up END frames.
+
+ * res/res_features.c: Don't try to send a parked call back to
+ itself. (closes issue #11622, reported by djrodman, patched by
+ me)
+
+ * main/autoservice.c: Don't store DTMF BEGIN frames while a channel
+ is in autoservice. It's just going to make ast_read() do a lot of
+ extra work when the channel comes back out of autoservice.
+ (closes issue #11628, patched by me)
+
+ * Makefile: List include/asterisk/version.h as a .PHONY target
+ because we want the commands listed for this target to be
+ executed regardless of whether the file exists or not. This fixes
+ having the version not up to date when running from svn. (closes
+ issue #11619, reported by plack, fixed by me)
+
+2007-12-25 02:27 +0000 [r94769] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: file says... build on the builders.
+
+2007-12-24 19:36 +0000 [r94763-94767] Tilghman Lesher <tlesher at digium.com>
+
+ * main/channel.c: Race: we need to wait to queue a NewChannel event
+ until after the channel is inserted into the channel list. The
+ reason is because some manager users immediately queue requests
+ from the channel when they see that event and are confused when
+ Asterisk reports no such channel. (Closes issue #11632)
+
+ * channels/chan_sip.c: More deadlock avoidance code (this time
+ between sip_monitor and sip_hangup) Reported by: apsaras Patch
+ by: tilghman (Closes issue #11413)
+
+ * channels/chan_sip.c: Another bit of bad logic in realtime_peer
+ Reported by: dimas Patch by: dimas (Closes issue #11631)
+
+2007-12-23 01:21 +0000 [r94660] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Argh... I suppose third time's the charm.
+
+2007-12-21 20:21 +0000 [r94468-94543] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Bunch of coding guidelines cleanup
+
+ * apps/app_voicemail.c: Better quota support for using IMAP storage
+ voicemail (closes issue #11415, reported by jaroth) (closes issue
+ #11152, reported by selsky) Patch provided by jaroth
+
+ * apps/app_voicemail.c: The mail_copy c-client function does not
+ expect a full imap mailbox string, just the name of the mailbox.
+ (closes issue #11419, reported and patched by jaroth, with
+ additional patchwork from me)
+
+ * main/dial.c: Since we are freeing list elements within a list
+ traversal, we need to use the safe traversal and remove the item
+ from the list before freeing it. (closes issue 11612, reported by
+ dtyoo)
+
+2007-12-21 16:37 +0000 [r94466] Russell Bryant <russell at digium.com>
+
+ * main/pbx.c, include/asterisk/pbx.h: Convert the contexts lock to
+ a read/write lock to resolve a deadlock. This has a nice side
+ benefit of improving performance. :) (closes issue #11609)
+ (closes issue #11080)
+
+2007-12-21 16:11 +0000 [r94420-94464] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Removing a debug message I accidentally just
+ committed
+
+ * main/say.c, apps/app_queue.c: Fixing Portuguese syntax for saying
+ dates and times. Also some coding guidelines cleanup. (closes
+ issue #11599, reported and patched by caio1982, coding guidelines
+ cleanup by me)
+
+2007-12-21 15:07 +0000 [r94418] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c: Fix for restart-as-user problem reported via the
+ -dev list
+
+2007-12-20 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.16.2 released.
+
+2007-12-20 20:22 +0000 [r94215-94256] Russell Bryant <russell at digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 94255 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) |
+ 5 lines Fix another potential seg fault ... (closes issue #11606)
+ Reported by: dimas ........
+
+ * channels/chan_zap.c: Fix a deadlock in d-channel handling in
+ chan_zap. This deadlock was introduced by the fix to ensure that
+ channels are properly locked when handling channel variables.
+ There were sections of this code where the channel pvt was locked
+ before the channel lock, when in fact it _must_ be the other way
+ around. (closes issue #11582) Reported by: bugi
+
+ * /: Blocked revisions 94214 via svnmerge ........ r94214 | russell
+ | 2007-12-20 11:29:11 -0600 (Thu, 20 Dec 2007) | 2 lines Fix a
+ couple of places where it's possible to dereference a NULL
+ pointer. ........
+
+2007-12-19 23:02 +0000 [r94122] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_monitor.c: Sox versions 13.0.0 and newer do not have
+ "soxmix" and instead use sox -m. res_monitor needs to use this if
+ the user does not have soxmix. (closes issue #11589, reported by
+ amessina, patch inspired by amessina but with a flourish from me)
+
+2007-12-19 22:48 +0000 [r94077] Russell Bryant <russell at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Check
+ for the existence of the soxmix application on the target
+ platform and have the result available in autoconfig.h. (part of
+ issue #11589)
+
+2007-12-19 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.16.1 released.
+
+2007-12-19 17:29 +0000 [r93955] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: Make the 1.4 builders happy, ensure var is
+ NULL.
+
+2007-12-19 17:04 +0000 [r93949] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_iax2.c: Avoid segfault in chan_iax when peer isn't
+ defined (Closes issue #11602)
+
+2007-12-18 22:42 +0000 [r93764] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: FreeBSD also does not have byte swap
+ functions. Issue 11586, patch by sobomax.
+
+2007-12-18 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.16 released.
+
+2007-12-18 18:45 +0000 [r93668-93676] Tilghman Lesher <tlesher at digium.com>
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 93667 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007)
+ | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........
+
+2007-12-18 17:02 +0000 [r93625] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Rework deadlock avoidance used in ast_write,
+ since it meant that agent channels which were being monitored had
+ one audio file recorded and one empty audio file saved. (closes
+ issue #11529, reported by atis patched by me)
+
+2007-12-17 22:56 +0000 [r93381-93420] Jason Parker <jparker at digium.com>
+
+ * main/translate.c: What was I thinking when I wrote this
+ masterpiece? -1 + 1 = 0.. who woulda thunk it?.
+
+2007-12-17 22:28 +0000 [r93377] Joshua Colp <jcolp at digium.com>
+
+ * main/utils.c: Do not try to access information about a lock when
+ printing out a trylock attempt. It is possible for the lock that
+ it references to no longer be valid. This would have caused
+ segfaults or deadlocks. (issue #BE-263) (closes issue #11080)
+ Reported by: callguy (closes issue #11100) Reported by: callguy
+
+2007-12-17 21:12 +0000 [r93336] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/time.h: Today is tomorrow's yesterday, and
+ yesterday's tomorrow is today, and tomorrow's tomorrow is the day
+ after tomorrow, so who cares if you recycle anyway? If this
+ confuses you, that's nothing compared to what this fixes. ;-)
+
+2007-12-17 19:53 +0000 [r93291] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: We need to create the directory for a
+ voicemail user even if they are using IMAP storage since
+ greetings are stored in the filesystem. (closes issue #11388,
+ reported by spditner, patch by me inspired by a patch by
+ spditner)
+
+2007-12-17 18:05 +0000 [r93250] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_zap.c: If a call is received with a called number
+ IE containing nothing go to the 's' extension. (closes issue
+ #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2007-12-17 07:21 +0000 [r93183] Kevin P. Fleming <kpfleming at digium.com>
+
+ * funcs/Makefile, codecs/Makefile, cdr/Makefile, pbx/Makefile,
+ res/Makefile, channels/Makefile, formats/Makefile: fix some
+ copy-and-paste leftovers
+
+2007-12-17 07:15 +0000 [r93182] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
+ apps/app_queue.c, channels/chan_iax2.c: Issue 11574: Add
+ dependencies on res_monitor and res_features. I wonder if
+ Asterisk can run at all without res_features. My guess is that
+ there's propably a lot of more modules and the core that depends
+ on it. Reported by: caio1982 (closes issue #11574)
+
+2007-12-17 06:44 +0000 [r93180] Kevin P. Fleming <kpfleming at digium.com>
+
+ * formats, Makefile, codecs/Makefile, funcs, apps/Makefile,
+ configure, cdr/Makefile, build_tools/prep_tarball, makeopts.in,
+ formats/Makefile, pbx, res, channels, funcs/Makefile, codecs,
+ include/asterisk/autoconfig.h.in, build_tools/make_version, apps,
+ configure.ac, Makefile.moddir_rules, build_tools/prep_moduledeps
+ (removed), res/Makefile, pbx/Makefile, cdr, channels/Makefile: In
+ http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
+ rizzo brought up some issues related to the way that the metadata
+ required for menuselect and the rest of the build system is
+ extracted from the source files. Since I had a few hours to kill
+ on an airplane today, I decided to improve this situation... so
+ now the system caches the extracted metadata and uses it to build
+ the menuselect 'tree' as much as it can. The result of this is
+ that when a single source file is changed, only the metadata for
+ that file needs to be extracted again, and the rest is used from
+ the cache files. I also reduced the number of forked processes
+ required to do the metadata extraction; it was actually possible
+ to do most of what we needed in the Makefiles themselves without
+ using any shell scripts at all! On my laptop, these changes
+ resulted in an 80% decrease in the time required for the
+ 'menuselect.makeopts' automatic check to occur after editing a
+ single source file. While doing this work I also cleaned up a few
+ minor things in the Makefiles, adding a check for 'awk' to the
+ configure script and changed all remaining places we use 'grep'
+ or 'awk' to use the ones found by the configure script, and
+ changed the 'prep_tarball' script to build the menuselect
+ metadata so that tarballs of Asterisk will include it and won't
+ require the user to wait while it is extracted after unpacking.
+
+2007-12-14 17:36 +0000 [r93000] Russell Bryant <russell at digium.com>
+
+ * main/config.c: There are a lot of existing systems that #include
+ non-existent files. So, to make the transition to treating this
+ as an error a bit less painless, just issue a huge error message
+ for now. Then, later, we can reinstate the code that treats it as
+ a failure. (Thanks to philippel for the feedback)
+
+2007-12-14 15:16 +0000 [r92937] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Up the length of the format on the SIP
+ channel since it can now be rather long. (closes issue #11552)
+ Reported by: francesco_r
+
+2007-12-14 15:05 +0000 [r92934] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: fixed the sequencing of WAITING_4DIGS
+ state setting and overlap_task thread starting.
+
+2007-12-14 15:01 +0000 [r92933] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Change help documentation to match actual behavior
+ (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman
+ (Closes issue #11548)
+
+2007-12-14 01:24 +0000 [r92875] Mark Michelson <mmichelson at digium.com>
+
+ * include/asterisk/lock.h: When compiling with DETECT_DEADLOCKS,
+ don't spam the CLI with messages about possible deadlocks.
+ Instead just print the intended single message every five
+ seconds. (closes issue 11537, reported and patched by dimas)
+
+2007-12-13 21:28 +0000 [r92815] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_zap.c: Properly initialize polarity statuses, so
+ that they are detected properly. Reported by: julianjm Patch by:
+ julianjm (Closes issue #10238)
+
+2007-12-13 20:13 +0000 [r92809] Jason Parker <jparker at digium.com>
+
+ * main/pbx.c: Make application help text a little more clear about
+ the use of extensions in a filename.
+
+2007-12-13 20:03 +0000 [r92803-92807] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Prevent another potential fd leak
+
+ * apps/app_voicemail.c: Prevent a possible fd leak.
+
+2007-12-13 00:11 +0000 [r92696] Jason Parker <jparker at digium.com>
+
+ * main/config.c, channels/chan_sip.c, channels/chan_h323.c,
+ channels/chan_iax2.c: If a typo is found in a config file, we
+ previous continued on with what was already loaded. We do not
+ want to do this (see bug below for details). This makes it so
+ that if a [ is found without a ], the entire config will fail,
+ and nothing in it will be loaded. Isue #10690.
+
+2007-12-12 22:00 +0000 [r92656] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: emit a warning message when we drop a G.729B
+ CNG frame destined for the transcoder
+
+2007-12-12 21:15 +0000 [r92617] Jason Parker <jparker at digium.com>
+
+ * apps/app_meetme.c: Don't increment user count until after name
+ has been recorded (if enabled). Issue 11048, tested by pep.
+
+2007-12-12 19:40 +0000 [r92556] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: resolve compiler warning
+
+2007-12-12 17:46 +0000 [r92510] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_features.c: Correctly detect where a dynamic feature was
+ activated. Before this patch, the channel which initiated the
+ bridge was always assumed to have been the one which activated
+ the dynamic feature. This patch corrects this. (closes issue
+ #11529, reported and patched by nic_bellamy)
+
+2007-12-12 16:52 +0000 [r92463] Tilghman Lesher <tlesher at digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac: Test
+ directly for the API that fixed AST-2007-026, to ensure that
+ older versions of PostgreSQL are no longer acceptable. (Closes
+ issue #11526)
+
+2007-12-12 16:08 +0000 [r92443] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Removing an unused variable.
+
+2007-12-11 19:51 +0000 [r92363] Joshua Colp <jcolp at digium.com>
+
+ * main/global_datastores.c: Fix potential memory leak with the
+ dialed interfaces list if another memory allocation fails.
+ (closes issue #11507) Reported by: eliel Patches:
+ global_datastores.c.patch uploaded by eliel (license 64)
+
+2007-12-11 17:42 +0000 [r92323] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fixing autofill to be more accurate.
+ Specifically, if calls ahead of the current caller were ringing
+ members (but not yet bridged) there could be available members
+ and waiting callers who would not get matched up. The member
+ availability checker was correctly determining the number of
+ available members in this scenario, but the queue itself did not
+ parallelly reflect this status on the pending calls. This commit
+ corrects the issue. (closes issue #11459, reported by
+ equissoftware, patched by me)
+
+2007-12-10 16:36 +0000 [r92204] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Add G729A as another possible payload name for G729.
+ Some devices use this instead of G729, which is perfectly normal
+ since the payload number itself is defined and can't be used by
+ anything else so the name doesn't matter that much. (closes issue
+ #11483) Reported by: revolution Patches: rtp.diff uploaded by
+ revolution (license 346)
+
+2007-12-10 16:29 +0000 [r92202] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: If there are no members in a queue, then the
+ loop where the datastore for detecting duplicate dialed numbers
+ will be skipped, meaning the datastore isn't created. This means
+ that when we try to free it, there's a crash. This stops that
+ crash from occurring. (closes issue #11499, reported by slavon,
+ patched by eliel)
+
+2007-12-10 16:13 +0000 [r92200] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: It is possible for nativeformats to contain
+ more then one codec, so print out multiple ones. (closes issue
+ #11366) Reported by: ovi
+
+2007-12-10 14:04 +0000 [r92158] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Avoid reinvite race situations with two
+ Asterisks trying to reinvite each other in 1.4 and trunk. This
+ patch implements support for the 491 error code that Asterisk 1.4
+ generates on situations where we get an incoming INVITE and
+ already has one in progress. Thanks to mavetju for reporting and
+ to Raj Jain for an excellent explanation of the problem. Patch by
+ myself. Tested with 8 Asterisk servers connected to each other in
+ a training network. Closes issue #10481
+
+2007-12-07 23:29 +0000 [r91890] Jason Parker <jparker at digium.com>
+
+ * main/dsp.c: We need to make sure we free the input frame if we
+ return a different frame in ast_dsp_process. Issue 11273, pointed
+ out by dimas, with a patch by eliel.
+
+2007-12-07 22:30 +0000 [r91870] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: even though Asterisk explicitly requests that
+ endpoints using G.729 do *not* use Annex B (silence detection and
+ comfort noise generation) some do anyway; the transcoder card
+ interface does not currently work properly with CNG frames, so
+ trim off the CNG before sending the data
+
+2007-12-07 21:24 +0000 [r91777-91830] Russell Bryant <russell at digium.com>
+
+ * main/utils.c: Make the lock protecting each thread's list of
+ locks it currently holds recursive. I think that this will fix
+ the situation where some people have said that "core show locks"
+ locks up the CLI. (related to issue #11080)
+
+ * include/asterisk/lock.h: Fix another bug in the DEBUG_THREADS
+ code. The ast_mutex_init() function had the mutex attribute
+ object marked as static. This means that multiple threads
+ initializing locks at the same time could step on each other and
+ end up with improperly initialized locks. (found when tracking
+ down locking issues related to issue #11080)
+
+ * include/asterisk/lock.h: I love fixing lock related errors in the
+ lock debugging code. That's about as ironic as it gets in
+ Asterisk programming land. Anyway, I spotted this bug while
+ trying to track down why systems are locking up and acting weird
+ in issue #11080. The mutex attribute object was marked as static
+ in this function when it should not have been.
+
+ * apps/app_dial.c: * Add channel locking around datastore
+ operations that expect the channel to be locked. * Document why
+ we don't record Local channels in the dialed interfaces list. *
+ Remove the dialed variable as it isn't needed. * Restructure some
+ code for clarity and coding guidelines stuff
+
+ * apps/app_queue.c: * Add channel locking around datastore
+ operations that expect the channel to be locked. * Document why
+ we don't record Local channels in the dialed interfaces list. *
+ Handle memory allocation failure. * Remove the dialed variable,
+ as it wasn't actually needed. * Tweak some formatting to conform
+ to coding guidelines.
+
+ * main/autoservice.c: * Add a bit more of a verbose comment as to
+ why a hangup frame needs to be queued up if autoservice gets a
+ NULL return from ast_read(). * Make the process of queueing the
+ hangup frame more efficient by putting the frame where it is
+ going to end up and avoiding some locking and extra memory
+ allocations and freeing.
+
+2007-12-07 15:39 +0000 [r91737] Mark Michelson <mmichelson at digium.com>
+
+ * main/autoservice.c: Hangups that happen during autoservice were
+ not processed appropriately. This is because a hangup actually
+ causes a NULL frame to be received, not a hangup frame. Queueing
+ a hangup if we receive a NULL frame during autoservice corrects
+ this problem (closes issue #11467, reported by jmls, patched by
+ me)
+
+2007-12-07 02:51 +0000 [r91675-91693] Russell Bryant <russell at digium.com>
+
+ * apps/app_dial.c: Don't unlock the dialed_interfaces list until
+ we're done messing with the iterator.
+
+ * apps/app_dial.c, apps/app_queue.c: Allow dialing local channels
+ from Queue() and Dial() again. There was a slight flaw in the
+ code to prevent call forwards from looping that caused this
+ problem. (related to issue #11486)
+
+ * apps/app_queue.c: Fix in an issue in the call forwarding handling
+ code that was causing crashes on every call into a queue. I'm not
+ entirely sure about the logic in this part of the code, so I want
+ to look at it some more tomorrow. However, this makes it safe and
+ keeps it from crashing. (closes issue #11486, reported by adamg,
+ patched by me)
+
+2007-12-07 00:52 +0000 [r91637] Tilghman Lesher <tlesher at digium.com>
+
+ * main/rtp.c: At the end of a call, when we're reporting, RTCP may
+ already be partially torn down, so check for NULL dereference
+ Reported by: blitzrage Patch by: tilghman (Closes issue #11450)
+
+2007-12-06 20:25 +0000 [r91541] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: IMAP storage did not honor the maxmsg
+ setting in voicemail.conf, and it also had the possibility of
+ crashing if a user had more than 256 messages in their voicemail.
+ This patch kills two birds with one stone by adding maxmsg
+ support and also setting a hard limit on the number of messages
+ at 255 so that the crashes cannot happen. (closes issue #11101,
+ reported by Skavin, patched by me)
+
+2007-12-06 19:11 +0000 [r91501] Russell Bryant <russell at digium.com>
+
+ * main/loader.c, include/asterisk/module.h: Add a new module flag
+ to indicate that a build sum is present. Modules built against
+ older Asterisk 1.4 headers will now load properly with just a
+ warning indicating that they are old and may cause problems.
+ (patch by paravoid)
+
+2007-12-06 16:49 +0000 [r91439-91450] Joshua Colp <jcolp at digium.com>
+
+ * main/udptl.c: Fix various in the udptl implementation. It could
+ return empty modem frames, have an incorrect sequence number on
+ packets, and display the wrong sequence number in the debug
+ messages. (closes issue #11228) Reported by: Cache Patches:
+ udptl-4.patch uploaded by dimas (license 88)
+
+ * channels/chan_sip.c: Add support for accepting and sending T.38
+ in the initial INVITE. (closes issue #9402) Reported by: thdei
+
+2007-12-06 12:54 +0000 [r91366] Olle Johansson <oej at edvina.net>
+
+ * main/loader.c, include/asterisk/logger.h, main/logger.c: Make
+ sure logger is reloaded at general reload in the cli. (Discovered
+ during Asterisk training in Portugal)
+
+2007-12-05 22:57 +0000 [r91273-91292] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Reverting extra stuff I didn't mean to
+ commit
+
+ * apps/app_voicemail.c, apps/app_dial.c: The 'G' option for Dial()
+ did not properly handle the case where only a label was provided.
+ This was due to the fact that the answering channel did not have
+ an extension set, so ast_parseable_goto would fail. This fix
+ eliminates the call to ast_parseable_goto on the answering
+ channel since it is a wasteful call. The answering channel and
+ the calling channel are both directed to the same extension and
+ context, just different priorities, so we can just copy the
+ values from the calling channel to the answering channel and
+ increment the answering channel's priority. (closes issue #11382,
+ reported by jon, patch by me with correction by jon)
+
+2007-12-05 21:38 +0000 [r91237] Tilghman Lesher <tlesher at digium.com>
+
+ * sounds/Makefile: Upgrade to the latest version of extra sounds
+
+2007-12-05 17:31 +0000 [r90967-91192] Russell Bryant <russell at digium.com>
+
+ * main/threadstorage.c: Make the lock in the threadstorage
+ debugging code untracked to avoid a deadlock on thread
+ destruction. (closes issue #11207) Reported by: ys Patches:
+ threadstorage.c.diff uploaded by ys (license 281) Also fixes an
+ open bug report: (closes issue #11446)
+
+ * main/utils.c: When DEBUG_THREADS is enabled, we only have the
+ details about who is holding a lock that we are waiting on for a
+ mutex, not rwlocks. This should fix the problem where people have
+ reported "core show locks" crashing sometimes.
+
+ * include/asterisk/lock.h: Fix some crashes in chan_iax2 that were
+ reported as happening on Mac systems. It turns out that the
+ problem was the Mac version of the ast_atomic_fetchadd_int()
+ function. The Mac atomic add function returns the _new_ value,
+ while this function is supposed to return the old value. So, the
+ crashes happened on unreferencing objects. If the reference count
+ was decreased to 1, ao2_ref() thought that it had been decreased
+ to zero, and called the destructor. However, there was still an
+ outstanding reference around. (closes issue #11176) (closes issue
+ #11289)
+
+ * include/asterisk/file.h, configure,
+ include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/compiler.h: Modify file.h to maintain API
+ compatibility with earlier versions. If a recent compiler is
+ being used, then a warning will show up for any modules still
+ using the old name "private" instead of "_private". (patch
+ suggested by paravoid)
+
+ * main/pbx.c: Make some changes to some additions I made recently
+ for doing channel autoservice when looking up extensions. This
+ code was added to handle the case where a dialplan switch was in
+ use that could block for a long time. However, the way that I
+ added it, it did this for all extension lookups. However, lookups
+ in the in-memory tree of extensions should _not_ take long enough
+ to matter. So, move the autoservice stuff to be only around
+ executing a switch.
+
+2007-12-04 17:28 +0000 [r90876] Jason Parker <jparker at digium.com>
+
+ * main/channel.c: If we fail to create a channel after allocating a
+ timing fd, we need to make sure to close it. Issue 11454, patch
+ by eliel.
+
+2007-12-04 05:29 +0000 [r90798] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_dial.c: Fix build issue on the build cluster.
+
+2007-12-03 23:50 +0000 [r90736-90753] Tilghman Lesher <tlesher at digium.com>
+
+ * include/asterisk/compat.h: Solaris requires the inclusion of
+ sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by:
+ snuffy,tilghman (Closes issue #11430)
+
+ * res/res_config_pgsql.c: If both dbhost and dbsock were not set, a
+ NULL deref could result Reported by: xrg Patch by: tilghman
+ (Closes issue #11387)
+
+2007-12-03 23:12 +0000 [r90735] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_dial.c, main/channel.c, main/global_datastores.c
+ (added), channels/chan_local.c, main/Makefile,
+ include/asterisk/channel.h, include/asterisk/global_datastores.h
+ (added), apps/app_queue.c: A big one... This is the merge of the
+ forward-loop branch. The main change here is that call-forwards
+ can no longer loop. This is accomplished by creating a datastore
+ on the calling channel which has a linked list of all devices
+ dialed. If a forward happens, then the local channel which is
+ created inherits the datastore. If, through this progression of
+ forwards and datastore inheritance, a device is attempted to be
+ dialed a second time, it will simply be skipped and a warning
+ message will be printed to the CLI. After the dialing has been
+ completed, the datastore is detached from the channel and
+ destroyed. This change also introduces some side effects to the
+ code which I shall enumerate here: 1. Datastore inheritance has
+ been backported from trunk into 1.4 2. A large chunk of code has
+ been removed from app_dial. This chunk is the section of code
+ which handles the call forward case after the channel has been
+ requested but before it has been called. This was removed because
+ call-forwarding still works fine without it, it makes the code
+ less error-prone should it need changing, and it made this set of
+ changes much less painful to just have the forwarding handled in
+ one place in each module. 3. Two new files, global_datastores.h
+ and .c have been added. These are necessary since the datastore
+ which is attached to the channel may be created and attached in
+ either app_dial or app_queue, so they need a common place to find
+ the datastore info. This approach was taken in case similar
+ datastores are needed in the future, there will be a common place
+ to add them.
+
+2007-12-03 22:06 +0000 [r90696] Jason Parker <jparker at digium.com>
+
+ * apps/app_meetme.c: Make sure we always close the conference fd if
+ we have an open one. Issue 11383, reported by markmhy, patch by
+ eliel.
+
+2007-12-03 20:59 +0000 [r90639] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_mgcp.c: Changing some bad logic when calculating
+ the interdigit timeout. (closes issue #11402, reported and
+ patched by eferro)
+
+2007-12-03 20:51 +0000 [r90607] Jason Parker <jparker at digium.com>
+
+ * res/res_features.c: Fix crash in ParkAndAnnounce application.
+ Issue #11436, reported by lytledd, patch by eliel.
+
+2007-12-03 20:05 +0000 [r90548-90588] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: Do not create a smoother for G723.1 frames, they need
+ to be left alone to their native 20/24 byte size.
+
+ * .cleancount, main/channel.c, include/asterisk/channel.h: Preserve
+ the indication currently playing on a channel when a masquerade
+ operation happens. (issue #BE-88)
+
+2007-12-03 18:20 +0000 [r90546] Jason Parker <jparker at digium.com>
+
+ * channels/chan_iax2.c: Only log debug messages if debug is
+ enabled. Closes issue #11416, patch by casper.
+
+2007-12-02 18:18 +0000 [r90470] Russell Bryant <russell at digium.com>
+
+ * apps/app_queue.c: The other day when I went through making
+ changes as a result of the ao2_link() change, I added some code
+ to set pointers to NULL after they were unreferenced. This
+ pointed out that in this place, the object was unreferenced
+ before the code was done using it. So, move the unref down a
+ little bit. (crash reported by jmls on IRC)
+
+2007-12-02 09:34 +0000 [r90432] Tilghman Lesher <tlesher at digium.com>
+
+ * main/autoservice.c: Clarify the return value on autoservice.
+ Specifically, if you started autoservice and autoservice was
+ already on, it would erroneously return an error. Reported by:
+ adiemus Patch by: dimas (Closes issue #11433)
+
+2007-11-30 19:26 +0000 [r90310-90348] Russell Bryant <russell at digium.com>
+
+ * main/astobj2.c, main/manager.c, include/asterisk/astobj2.h,
+ apps/app_queue.c, channels/chan_iax2.c: Change the behavior of
+ ao2_link(). Previously, in inherited a reference. Now, it
+ automatically increases the reference count to reflect the
+ reference that is now held by the container. This was done to be
+ more consistent with ao2_unlink(), which automatically releases
+ the reference held by the container. It also makes it so it is no
+ longer possible for a pointer to be invalid after ao2_link()
+ returns.
+
+ * include/asterisk/astobj2.h: Add some notes on the behavior of
+ ao2_unlink() after a discussion with Tilghman
+
+2007-11-30 14:43 +0000 [r90269] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Fix locking issues under one legged replaces
+ scenarios. (closes issue #11420) Reported by: irroot Patches:
+ chan_sip_oneleg.patch uploaded by irroot (license 52)
+
+2007-11-30 00:16 +0000 [r90231] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_mgcp.c: Clear the DTMF buffer if the call times
+ out. (closes issue #11418, reported and patched by eferro)
+
+2007-11-29 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.15 released.
+
[... 13757 lines stripped ...]
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