[asterisk-commits] russell: tag 1.4.20-rc1 r114927 - /tags/1.4.20-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu May 1 11:58:26 CDT 2008
Author: russell
Date: Thu May 1 11:58:26 2008
New Revision: 114927
URL: http://svn.digium.com/view/asterisk?view=rev&rev=114927
Log:
Importing files for 1.4.20-rc1 release
Added:
tags/1.4.20-rc1/.lastclean (with props)
tags/1.4.20-rc1/.version (with props)
tags/1.4.20-rc1/ChangeLog (with props)
Added: tags/1.4.20-rc1/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.20-rc1/.lastclean?view=auto&rev=114927
==============================================================================
--- tags/1.4.20-rc1/.lastclean (added)
+++ tags/1.4.20-rc1/.lastclean Thu May 1 11:58:26 2008
@@ -1,0 +1,1 @@
+32
Propchange: tags/1.4.20-rc1/.lastclean
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.20-rc1/.lastclean
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.20-rc1/.lastclean
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.20-rc1/.version
URL: http://svn.digium.com/view/asterisk/tags/1.4.20-rc1/.version?view=auto&rev=114927
==============================================================================
--- tags/1.4.20-rc1/.version (added)
+++ tags/1.4.20-rc1/.version Thu May 1 11:58:26 2008
@@ -1,0 +1,1 @@
+1.4.20-rc1
Propchange: tags/1.4.20-rc1/.version
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: tags/1.4.20-rc1/.version
------------------------------------------------------------------------------
svn:keywords = none
Propchange: tags/1.4.20-rc1/.version
------------------------------------------------------------------------------
svn:mime-type = text/plain
Added: tags/1.4.20-rc1/ChangeLog
URL: http://svn.digium.com/view/asterisk/tags/1.4.20-rc1/ChangeLog?view=auto&rev=114927
==============================================================================
--- tags/1.4.20-rc1/ChangeLog (added)
+++ tags/1.4.20-rc1/ChangeLog Thu May 1 11:58:26 2008
@@ -1,0 +1,17361 @@
+2008-05-01 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.20-rc1 released.
+
+2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+ Merge changes from team/russell/iax2_find_callno and
+ iax2_find_callno_1.4 These changes address a critical performance
+ issue introduced in the latest release. The fix for the latest
+ security issue included a change that made Asterisk randomly
+ choose call numbers to make them more difficult to guess by
+ attackers. However, due to some inefficient (this is by far, an
+ understatement) code, when Asterisk chose high call numbers,
+ chan_iax2 became unusable after just a small number of calls. On
+ a small embedded platform, it would not be able to handle a
+ single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
+ more than about 16 IAX2 channels. Ouch. These changes address
+ some performance issues of the find_callno() function that have
+ bothered me for a very long time. On every incoming media frame,
+ it iterated through every possible call number trying to find a
+ matching active call. This involved a mutex lock and unlock for
+ each call number checked. So, if the random call number chosen
+ was 20000, then every media frame would cause 20000 locks and
+ unlocks. Previously, this problem was not as obvious since
+ Asterisk always chose the lowest call number it could. A second
+ container for IAX2 pvt structs has been added. It is an astobj2
+ hash table. When we know the remote side's call number, the pvt
+ goes into the hash table with a hash value of the remote side's
+ call number. Then, lookups for incoming media frames are a very
+ fast hash lookup instead of an absolutely insane array traversal.
+ In a quick test, I was able to get more than 3600% more IAX2
+ channels on my machine with these changes.
+
+2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
+ in Huntsville together with Mark M (putnopvut) (closes issue
+ #12363) Reported by: jvandal Tested by: putnopvut, oej
+
+2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
+ for indexing through the iaxs/iaxsl arrays so that the size of
+ the arrays can be adjusted in one place, and change the size of
+ the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
+ defined
+
+ * Makefile.rules: pay attention to *all* header files for
+ dependency tracking, not just the local ones (inspired by r578 of
+ asterisk-addons by tilghman)
+
+2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
+ variables instead of the channel's macrocontext and macroexten
+ fields. This is needed because if macros are daisy-chained, the
+ incorrect context and extension are placed on the new channel. I
+ also added locking to the channel prior to accessing these
+ variables as noted in trunk's janitor project file. (closes issue
+ #12549) Reported by: darren1713 Patches:
+ app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
+ (with modifications from me) Tested by: putnopvut
+
+2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker at digium.com>
+
+ * res/res_config_pgsql.c: Change warning message to debug, since
+ there are cases where 0 results is perfectly fine.
+
+2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
+ 2008) | 2 lines stop script from appending source code if run
+ multiple times ........
+
+2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
+ embedded, they take on a different name, without the ".so"
+ extension. Specifically check for this name, when we're checking
+ if a module is loaded. (Closes issue #12534)
+
+2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright at gmail.com>
+
+ * configure, configure.ac: When we don't explicitly pass a path to
+ the --with-tds configure option, we may end up finding tds.h in
+ /usr/local/include instead of /usr/include. If this happens, the
+ grep that looks for the version (from tdsver.h) will fail and
+ we'll have some problems during the build.
+
+2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/scripts/vmail.cgi: Clicking forward without selecting a
+ message leaves an errant .lock file. (closes issue #12528)
+ Reported by: pukepail Patches: patch.diff uploaded by pukepail
+ (license 431)
+
+2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Use consistent logic for checking to see if
+ a call number has been chosen yet. Also, remove some redundant
+ logic I recently added in a fix.
+
+2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c: Move the unlock of the spyee channel to
+ outside the start_spying() function so that the channel is not
+ unlocked twice when using whisper mode.
+
+2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher at digium.com>
+
+ * configs/zapata.conf.sample, configs/iax.conf.sample,
+ configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
+ documentation files that actually exist. (closes issue #12516)
+ Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
+ by linuxmaniac (license 472)
+
+2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
+ for instance, an AMI redirect of two channels which are natively
+ bridged will preserve audio on both channels. This prevents a
+ problem with Asterisk not re-inviting due to one of the channels
+ having being a zombie. (closes issue #12513) Reported by:
+ mneuhauser Patches:
+ asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
+ mneuhauser (license 425)
+
+ * apps/app_queue.c: Output of channel variables when
+ eventwhencalled=vars was set was being truncated two characters.
+ This patch corrects the problem. (closes issue #12493) Reported
+ by: davidw
+
+ * channels/chan_local.c: Resolve a deadlock in chan_local by
+ releasing the channel lock temporarily. (closes issue #11712)
+ Reported by: callguy Patches: 11712.patch uploaded by putnopvut
+ (license 60) Tested by: acunningham
+
+2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_local.c: Ensure that when we set the accountcode,
+ it actually shows up in the CDR. (Fix for AMI Originate) (Closes
+ issue #12007)
+
+2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: Fix a silly mistake in a change I made
+ yesterday that caused chan_iax2 to blow up very quickly. (issue
+ #12515)
+
+2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Only have one max-forwards header in
+ outbound REFERs. Discovered in the Asterisk SIP Masterclass in
+ Orlando. Thanks Joe!
+
+2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell at digium.com>
+
+ * main/http.c: Improve some broken cookie parsing code. Previously,
+ manager login over HTTP would only work if the mansession_id
+ cookie was first. Now, the code builds a list of all of the
+ cookies in the Cookie header. This fixes a problem observed by
+ users of the Asterisk GUI. (closes AST-20)
+
+ * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
+ the correct next channel to not always work. Also, remove setting
+ the amount of time to wait for a digit from 5 seconds back down
+ to 1/10 of a second. I believe this was so the beep didn't get
+ played over and over really fast, but a while back I put in
+ another fix for that issue. (closes issue #12498) Reported by:
+ jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
+ jsmith (license 15)
+
+2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker at digium.com>
+
+ * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
+ module. (closes issue #11575) Reported by: sunder Patches:
+ M11575_14_rev3.diff uploaded by junky (license 177)
+ bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell at digium.com>
+
+ * main/manager.c, include/asterisk/manager.h: Store the manager
+ session ID explicitly as 4 byte ID instead of a ulong. The
+ mansession_id cookie is coded to be limited to 8 characters of
+ hex, and this could break logins from 64-bit machines in some
+ cases. (inspired by AST-20)
+
+ * channels/chan_iax2.c: Fix find_callno_locked() to actually return
+ the callno locked in some more cases.
+
+2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Add 502 support for both directions, not
+ only one... (see r114571)
+
+2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp at digium.com>
+
+ * main/pbx.c: Instead of stopping dialplan execution when SayNumber
+ attempts to say a large number that it can not print out a
+ message informing the user and continue on. (closes issue #12502)
+ Reported by: bcnit
+
+2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
+ to processing a response code
+
+2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell at digium.com>
+
+ * channels/chan_iax2.c: When we receive a full frame that is
+ supposed to contain our call number, ensure that it has the
+ correct one. (closes issue #10078) (AST-2008-006)
+
+ * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
+ thought I was going to be able to leave 1.4 alone, but that was
+ not the case. I ran into some problems with G.722 in 1.4, so I
+ have merged in all of the fixes in this area that I have made in
+ trunk/1.6.0, and things are happy again.
+
+ * res/res_musiconhold.c: Trivial change to read the number of
+ samples from a frame before calling ast_write()
+
+ * res/res_features.c: After a parked call times out, allow the call
+ back to the parker to time out. (closes issue #10890)
+
+ * channels/chan_iax2.c: If the dial string passed to the call
+ channel callback does not indicate an extension, then consider
+ the extension on the channel before falling back to the default.
+ (closes issue #12479) Reported by: darren1713 Patches:
+ exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
+ 116)
+
+ * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
+ team/russell/issue_9520 These changes make sure that the
+ reference count for sip_peer objects properly reflects the fact
+ that the peer is sitting in the scheduler for a scheduled
+ callback for qualifying peers or for expiring registrations.
+ Without this, it was possible for these callbacks to happen at
+ the same time that the peer was being destroyed. This was
+ especially likely to happen with realtime peers, and for people
+ making use of the realtime prune CLI command. (closes issue
+ #9520) Reported by: kryptolus Committed patch by me
+
+2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Only drop audio if we receive it without a
+ progress indication. We allow other frames through such as DTMF
+ because they may be needed to complete the call. (closes issue
+ #12440) Reported by: aragon
+
+2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_playback.c: Ensure that help text terminates with a
+ newline
+
+ * res/res_musiconhold.c: MOH usage information needs a terminating
+ newline, or else "asterisk -rx 'help moh reload'" will hang.
+ Reported via -dev list, fixed by me.
+
+2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell at digium.com>
+
+ * main/manager.c: Don't destroy a manager session if poll() returns
+ an error of EAGAIN.
+
+ * Makefile: ensure directories are created before we try to install
+ stuff into them
+
+ * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
+ bininstall
+
+2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_zap.c, main/callerid.c: Clearing up error messages
+ so they make a bit more sense. Also removing a redundant error
+ message. Issue AST-15
+
+2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell at digium.com>
+
+ * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
+ (closes issue #12476) Reported by: davidw Patch by me
+
+2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright at gmail.com>
+
+ * channels/chan_sip.c: Only complete the SIP channel name once for
+ 'sip show channel <channel>'
+
+2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_setcallerid.c: For consistency sake, ensure that the
+ values that ${CALLINGPRES} returns are valid as an input to
+ SetCallingPres. (Closes issue #12472)
+
+2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell at digium.com>
+
+ * main/autoservice.c: Remove redundant safety net. The check for
+ the autoservice channel list state accomplishes the same goal in
+ a better way. (issue #12470) Reported By: atis
+
+2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_chanspy.c: Declaration of the peer channel in this scope
+ was making it so the peer variable defined in the outer scope was
+ never set properly, therefore making iterating through the
+ channel list always restart from the beginning. This bug would
+ have affected anyone who called chanspy without specifying a
+ first argument. (closes issue #12461) Reported by: stever28
+
+ * main/frame.c, include/asterisk/dsp.h: Add prototype for
+ ast_dsp_frame_freed. I'm not sure how this was compiling
+ before...
+
+ * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
+ possible for a reference to a frame which was part of a freed DSP
+ to still be referenced, leading to memory corruption and eventual
+ crashes. This code change ensures that the dsp is freed when we
+ are finished with the frame. This change is very similar to a
+ change Russell made with translators back a month or so ago.
+ (closes issue #11999) Reported by: destiny6628 Patches:
+ 11999.patch uploaded by putnopvut (license 60) Tested by:
+ destiny6628, victoryure
+
+2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell at digium.com>
+
+ * Makefile: Fix the bininstall target to install from subdirs, as
+ well. (closes issue AST-8, patch from bmd at switchvox)
+
+2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * res/res_jabber.c: Use keepalives effectively in order diagnose
+ bug #12432.
+
+2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_agi.c: Add special case for when the agi cannot be
+ executed, to comply with the documentation that we return failure
+ in that case. (closes issue #12462) Reported by: fmueller
+ Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
+ (license 14) Tested by: fmueller
+
+2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright at gmail.com>
+
+ * apps/app_chanspy.c: Make sure we have enough room for the
+ recording's filename.
+
+2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
+ initializing the structure to all zeroes in case it contains
+ fields that we don't write values into (which it does as of
+ Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
+
+2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
+ to 1.4 (Closes issue #12457)
+
+2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
+ options in a previously configured user don't override settings
+ in general. (closes issue #12458) Reported by: tzafrir Patches:
+ chanzap_users_sections.diff uploaded by tzafrir (license 46)
+
+2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Include the proper headers for using mkdir on
+ FreeBSD. (closes issue #12430) Reported by: ys Patches:
+ app_meetme.c.diff uploaded by ys (license 281)
+
+2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Handle subscribe queues in all situations...
+ Thanks to festr_ on irc for telling me about this bug.
+
+2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker at digium.com>
+
+ * contrib/scripts/autosupport: Update Digium autosupport script,
+ for more useful information. (closes issue #12452) Reported by:
+ angler Patches: autosupport.diff uploaded by angler (license 106)
+
+ * apps/app_queue.c: Allow autofill to work in the general section
+ of queues.conf. Additionally, don't try to (re)set options when
+ they have empty values in realtime (all unset columns would have
+ an empty value). (closes issue #12445) Reported by: atis Patches:
+ 12445-autofill.diff uploaded by qwell (license 4)
+
+ * channels/chan_h323.c: The call_token on the pvt can occasionally
+ be NULL, causing a crash. If it is NULL, we can skip this
+ channel, since it can't the one we're looking for. (closes issue
+ #9299) Reported by: vazir
+
+2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Increase the retry count when attempting to show
+ channels. This apparently cleared an issue someone was seeing
+ when attempting to show channels when the load was high. (closes
+ issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
+ by russell (license 2) Tested by: falves11
+
+ * apps/app_dial.c, apps/app_queue.c: If the datastore has been
+ moved to another channel due to a masquerade, then freeing the
+ datastore here causes an eventual double free when the new
+ channel hangs up. We should only free the datastore if we were
+ able to successfully remove it from the channel we are
+ referencing (i.e. the datastore was not moved). (closes issue
+ #12359) Reported by: pguido
+
+ * main/channel.c: Save a local copy of the generate callback prior
+ to unlocking the channel in case the generate callback goes NULL
+ on us after the channel is unlocked. Thanks to Russell for
+ pointing this need out to me.
+
+2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: It is possible for the remote side to say
+ they want T38 but not give any capabilities. (closes issue
+ #12414) Reported by: MVF
+
+ * main/rtp.c: Don't change the SSRC when a new source comes into
+ play, this might happen quite often and depending on the remote
+ side... they might not like this. (closes issue #12353) Reported
+ by: dimas
+
+2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_iax2.c: Several places in the code called
+ find_callno() (which releases the lock on the pvt structure) and
+ then immediately locked the call and did things with it.
+ Unfortunately, the call can disappear between the find_callno and
+ the lock, causing Bad Stuff(tm) to happen. Added
+ find_callno_locked() function to return the callno withtout
+ unlocking for instances that it is needed. (issue #12400)
+ Reported by: ztel
+
+2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker at digium.com>
+
+ * main/pbx.c: It's possible that a channel can have an async goto
+ on the successful execution of an application as well. Closes
+ issue #12172.
+
+2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson at digium.com>
+
+ * res/res_features.c: Fix a race condition that may happen between
+ a sip hangup and a "core show channel" command. This patch adds
+ locking to prevent the resulting crash. (closes issue #12155)
+ Reported by: tsearle Patches: show_channels_crash2.patch uploaded
+ by tsearle (license 373) Tested by: tsearle
+
+ * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
+ LOW_MEMORY is enabled.
+
+ * channels/chan_sip.c: Be sure that we're not about to set
+ bridgepvt NULL prior to dereferencing it. (closes issue #11775)
+ Reported by: fujin
+
+2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker at digium.com>
+
+ * main/file.c: Only try to prefix language if we are not using an
+ absolute path (suffix it otherwise).
+ en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
+ issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
+ uploaded by qwell (license 4) Tested by: kuj, qwell
+
+2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
+ fix. (closes issue #12404) Reported by: IgorG Patches:
+ voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
+
+ * apps/app_meetme.c: Create the directory where name recordings
+ will go if it does not exist. (closes issue #12311) Reported by:
+ rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
+
+ * channels/chan_sip.c: Don't add custom URI options if they don't
+ exist OR they are empty. (closes issue #12407) Reported by:
+ homesick Patches: uri_options-1.4.diff uploaded by homesick
+ (license 91)
+
+2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: We need to set the persistant_route [sic]
+ parameter for the sip_pvt during the initial INVITE, no matter if
+ we're building the route set from an INVITE request or response.
+ (closes issue #12391) Reported by: benjaminbohlmann Tested by:
+ benjaminbohlmann
+
+2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher at digium.com>
+
+ * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
+ not exist in cdr.conf, then an unload/load sequence is needed to
+ correct the problem. Track whether the load succeeded with a
+ variable, so we can fix this with a simple reload event, instead.
+
+2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
+ server and we are unable to reply (for example they have a secret
+ configured, but we do not) then queue a hangup frame on the
+ Asterisk channel. This will cause the channel to hangup and a
+ HANGUP to be sent via IAX2 to the remote side which is the proper
+ thing to do in this scenario. (closes issue #12385) Reported by:
+ viraptor
+
+2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
+ a reinvite), then we should not send a BYE. (closes issue #12392)
+ Reported by: fnordian Patches: chan_sip.patch uploaded by
+ fnordian (license 110) with small modification from me
+
+2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson at digium.com>
+
+ * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
+ happy
+
+2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_parkandannounce.c: Fix potential buffer overflow that
+ could happen if more than 100 announce files were specified when
+ calling ParkAndAnnounce. This overflow is not exploitable
+ remotely and so there is no need for a security advisory. (closes
+ issue #12386) Reported by: davidw
+
+2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c: Add a little more that is required for
+ previously added devices.
+
+ * channels/chan_skinny.c: Add support for several new(ish) devices
+ - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
+ for providing me the required information.
+
+ * main/asterisk.c: Work around some silliness caused by
+ sys/capability.h - this should fix compile errors a number of
+ users have been experiencing.
+
+2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher at digium.com>
+
+ * contrib/scripts/astgenkey.8: Add security note on astgenkey's
+ manpage. (closes issue #12373) Reported by: lmamane Patches:
+ 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
+
+ * channels/chan_sip.c: Move check for still-bridged channels out a
+ little further, to avoid possible deadlocks. (Closes issue
+ #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: callguy
+
+2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/slinfactory.h, main/slinfactory.c,
+ main/audiohook.c: If audio suddenly gets fed into one side of a
+ channel after a lapse of frames flush the other factory so that
+ old audio does not remain in the factory causing the sync code to
+ not execute. (closes issue #12296) Reported by: jvandal
+
+2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
+ fixes a for loop (in realtime_peer) to check all the
+ ast_variables the loop was intending to test rather than just the
+ first one. The change exposed the problem of calling memcpy on a
+ NULL pointer, in this case the passed in sockaddr_in struct which
+ is now checked.
+
+2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker at digium.com>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
+ playback with noanswer (and add earlyrtp option). (closes issue
+ #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
+ (license 30) Tested by: pj, qwell, DEA, wedhorn
+
+2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher at digium.com>
+
+ * funcs/func_strings.c: Force ast_mktime() to check for DST, since
+ strptime(3) does not. (Closes issue #12374)
+
+2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: This fix prevents a deadlock that was experienced
+ in chan_local. There was deadlock prevention in place in
+ chan_local, but it would not work in a specific case because the
+ channel was recursively locked. By unlocking the channel prior to
+ calling the generator's generate callback in
+ ast_read_generator_actions(), we prevent the recursive locking,
+ and therefore the deadlock. (closes issue #12307) Reported by:
+ callguy Patches: 12307.patch uploaded by putnopvut (license 60)
+ Tested by: callguy
+
+2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
+ Reported by: vinsik Tested by: tecnoxarxa This one line change
+ makes an if inside a for loop (in realtime_peer) check all the
+ ast_variables the loop was intending to test rather than just the
+ first one.
+
+2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * channels/chan_gtalk.c: Free newly allocated channel before
+ returning
+
+ * channels/chan_gtalk.c: Prevent call connections when codecs don't
+ match. (closes issue #10604) Reported by: keepitcool Patches:
+ branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
+ by: phsultan
+
+2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp at digium.com>
+
+ * main/Makefile: Pass in the path to Zaptel for systems that
+ install Zaptel headers in a separate location.
+
+ * main/asterisk.c: One thing at a time... let's get 1.4 building.
+
+2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard at digium.com>
+
+ * main/asterisk.c: add a Zaptel timer check to verify the timer is
+ responding when Zaptel support is compiled into Asterisk and
+ Zaptel drivers are loaded. This will help people not waste their
+ valuable time debugging side effects.
+
+2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_zap.c: Fix the testing of the "res" variable so
+ that it is more logically correct and makes the correct warning
+ and debug messages print. (closes issue #12361) Reported by:
+ one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
+ (license 23)
+
+ * main/manager.c: Fix a race condition in the manager. It is
+ possible that a new manager event could be appended during a
+ brief time when the manager is not waiting for input. If an event
+ comes during this period, we need to set an indicator that there
+ is an event pending so that the manager doesn't attempt to wait
+ forever for an event that already happened. (closes issue #12354)
+ Reported by: bamby Patches: manager_race_condition.diff uploaded
+ by bamby (license 430) (comments added by me)
+
+ * apps/app_queue.c: Ensure that there is no timeout if none is
+ specified. (closes issue #12349) Reported by: johnlange
+
+2008-04-01 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.19 released.
+
+2008-03-28 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.19-rc4 released.
+
+2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker at digium.com>
+
+ * formats/format_wav_gsm.c: The file size of WAV49 does not need to
+ be an even number. (closes issue #12128) Reported by: mdu113
+ Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
+ Tested by: qwell, mdu113
+
+2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher at digium.com>
+
+ * doc/valgrind.txt: Update debugging text, since Valgrind
+ eliminated the --log-file-exactly option. (Closes issue #12320)
+
+ * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
+ NULL. (closes issue #12300) Reported by: festr Patches:
+ acl.c.patch uploaded by festr (license 443)
+
+2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf at digium.com>
+
+ * apps/app_playback.c, main/pbx.c: These small documentation
+ updates made in response to a query in asterisk-users, where a
+ user was using Playback, but needed the features of Background,
+ and had no idea that Background existed, or that it might provide
+ the features he needed. I thought the best way to avert these
+ kinds of queries was to provide "See Also" references in all
+ three of "Background", "Playback", "WaitExten". Perhaps a project
+ to do this with all related apps is in order.
+
+ * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
+ #12302) Reported by: pj Tested by: murf These changes will set a
+ channel variable ~~EXTEN~~ just before generating code for a
+ switch, with the value of ${EXTEN}. The exten is marked as having
+ a switch, and ever after that, till the end of the exten, we
+ substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
+ arguments; (and the ${EXTEN: also). The reason for this, is that
+ because switches are coded using separate extensions to provide
+ pattern matching, and jumping to/from these switch extensions
+ messes up the ${EXTEN} value, which blows the minds of users.
+
+2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker at digium.com>
+
+ * main/frame.c: Put this flag back so we don't change the API.
+
+ * main/frame.c: Remove excessive smoother optimization that was
+ causing audio glitches (small "pops") after (about 200ms later)
+ an "incorrectly" sized frame was received. While it would be very
+ nice to keep this as optimized as possible, it makes no sense for
+ the smoother to be dropping random bits of audio like this. Isn't
+ that the whole point of a smoother? Closes issue #12093.
+
+2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp at digium.com>
+
+ * contrib/scripts/autosupport: Update autosupport script. (closes
+ issue #12310) Reported by: angler Patches: autosupport.diff
+ uploaded by angler (license 106)
+
+2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming at digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
+ 2008) | 2 lines update UPGRADE notes to document usage of the
+ script ........
+
+2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: This code change is made just for
+ clarification. It does exactly the same thing as before. It just
+ doesn't look as wrong.
+
+ * apps/app_voicemail.c: Add a lock to the vm_state structure and
+ use the lock around mail_open calls to prevent concurrent access
+ of the same mailstream. This, along with trunk's ability to
+ configure TCP timeouts for IMAP storage will help to prevent
+ crashes and hangs when using voicemail with IMAP storage. (closes
+ issue #10487) Reported by: ewilhelmsen
+
+2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
+ Merged revisions 111019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
+ 2008) | 2 lines add a script to make getting the iLBC source code
+ simple for end users ........
+
+2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: If we are requested to authenticate a
+ reinvite make sure that it contains T38 SDP if need be. (closes
+ issue #11995) Reported by: fall
+
+ * channels/chan_iax2.c: Make sure that full video frames are sent
+ whenever the 15 bit timestamp rolls over. (closes issue #11923)
+ Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
+ mihai (license 94)
+
+2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming at digium.com>
+
+ * UPGRADE.txt: add note that the user will need to enable
+ codec_ilbc to get it to build
+
+ * codecs/ilbc/StateConstructW.h (removed),
+ codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
+ (removed), codecs/ilbc/getCBvec.c (removed),
+ codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
+ (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
+ (removed), codecs/ilbc/getCBvec.h (removed),
+ codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
+ (removed), codecs/ilbc/FrameClassify.c (removed),
+ codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
+ codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
+ (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
+ (removed), codecs/ilbc/anaFilter.c (removed),
+ codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
+ (removed), codecs/ilbc/doCPLC.h (removed),
+ codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
+ codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
+ (removed), codecs/ilbc/createCB.h (removed), CHANGES,
+ codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
+ (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
+ codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
+ codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
+ (removed), codecs/ilbc/iCBSearch.h (removed),
+ codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
+ codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
+ (removed), codecs/ilbc/hpOutput.h (removed),
+ codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
+ codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
+ (removed), codecs/ilbc/iCBConstruct.c (removed),
+ codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
+ (removed), codecs/ilbc/syntFilter.h (removed),
+ codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
+ (removed): Merged revisions 110869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
+ 2008) | 2 lines due to licensing restrictions, we cannot
+ distribute the source code for iLBC encoding and decoding... so
+ remove it, and add instructions on how the user can obtain it
+ themselves ........
+
+2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker at digium.com>
+
+ * cdr/cdr_custom.c: Make file access in cdr_custom similar to
+ cdr_csv. Fixes issue #12268. Patch borrowed from r82344
+
+2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler at digium.com>
+
+ * channels/chan_sip.c: This one line change makes an if inside a
+ for loop (in realtime_peer) check all the ast_variables the loop
+ was intending to test rather than just the first one.
+
+2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: When reverting a commit, I accidentally left
+ in this bit which was an experiment to see what would happen. It
+ passed the compile test, and I didn't notice I had left this
+ change in too. So this is a revert of a revert...sort of.
+
+2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ main/app.c: Add an option (transmit_silence) which transmits
+ silence during both Record() and DTMF generation. The reason this
[... 16559 lines stripped ...]
More information about the asterisk-commits
mailing list