[asterisk-commits] russell: tag 1.4.18 r100735 - in /tags/1.4.18: .lastclean .version ChangeLog
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jan 28 17:58:16 CST 2008
Author: russell
Date: Mon Jan 28 17:58:16 2008
New Revision: 100735
URL: http://svn.digium.com/view/asterisk?view=rev&rev=100735
Log:
Importing files for 1.4.18 release
Added:
tags/1.4.18/.lastclean (with props)
tags/1.4.18/.version (with props)
tags/1.4.18/ChangeLog (with props)
Added: tags/1.4.18/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.4.18/.lastclean?view=auto&rev=100735
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--- tags/1.4.18/ChangeLog (added)
+++ tags/1.4.18/ChangeLog Mon Jan 28 17:58:16 2008
@@ -1,0 +1,15298 @@
+2008-01-28 Russell Bryant <russell at digium.com>
+
+ * Asterisk 1.4.18 released.
+
+2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher at digium.com>
+
+ * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
+ (went to 't' instead of 'T')
+
+2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
+ chan_vpb. It is alive and well.
+
+2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker at digium.com>
+
+ * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
+ greeting files into the database like we do with the others.
+ Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
+ by dimas (license 88)
+
+2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: For some reason, the use of this strdupa()
+ is leading to memory corruption on freebsd sparc64. This trivial
+ workaround fixes it. (closes issue #10300, closes issue #11857,
+ reported by mattias04 and Home-of-the-Brave)
+
+ * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
+ #11342) Reported by: DEA Patches: res_features-park.txt uploaded
+ by DEA (license 3)
+
+2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker at digium.com>
+
+ * channels/chan_zap.c: Correct a comment which made little/no
+ sense.
+
+2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell at digium.com>
+
+ * main/channel.c, channels/chan_local.c,
+ include/asterisk/channel.h: Make some deadlock related fixes.
+ These bugs were discovered and reported internally at Digium by
+ Steve Pitts. - Fix up chan_local to ensure that the channel lock
+ is held before the local pvt lock. - Don't hold the channel lock
+ when executing the timing function, as it can cause a deadlock
+ when using chan_local. This actually changes the code back to be
+ how it was before the change for issue #10765. But, I added some
+ other locking that I think will prevent the problem reported
+ there, as well.
+
+2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher at digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
+ channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
+ channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
+ pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
+ the scheduler, ignoring the return value could possibly cause
+ memory to be accessed after it is freed, which causes all sorts
+ of random memory corruption. Instead, if a deletion fails, wait a
+ bit and try again (noting that another thread could change our
+ taskid value). (closes issue #11386) Reported by: flujan Patches:
+ 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, flujan, stuarth`
+
+2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
+ now preferred that users of Voicetronix products use chan_zap in
+ combination with their zaptel drivers.
+
+2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker at digium.com>
+
+ * channels/chan_sip.c: This would have never been true, since we're
+ passing (sizeof(req.data) - 1) as the len to recvfrom().
+
+2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming at digium.com>
+
+ * include/asterisk/app.h: make these macros not assume that the
+ only other field in the structure is 'argc'... this is true when
+ someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
+ to define your own structure as long as it has the right fields
+
+2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell at digium.com>
+
+ * main/asterisk.c: Update main Asterisk copyright info to 2008
+
+2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker at digium.com>
+
+ * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
+ Patches: bug-11832.diff uploaded by snuffy (license 35)
+
+2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Second attempt. Don't change invitestate
+ when receiving 18x messages in CANCEL state. (issue #11736)
+ Reported by: MVF Patch by oej.
+
+ * channels/chan_sip.c: Make sure we don't cancel destruction on
+ calls in CANCEL state, even if we get 183 while waiting for
+ answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
+ bug11736.txt uploaded by oej (license 306) Tested by: MVF
+
+2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_externalivr.c: Fixing a typo.
+
+2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell at digium.com>
+
+ * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
+ beginning of a list of channels to potentially spy on. However,
+ if there were no matching channels, it would beep at you over and
+ over, which is pretty annoying. Now, it will only beep once in
+ the case that there are no channels to spy on, but it will still
+ beep again once it reaches the beginning of the channel list
+ again. (closes issue #11738, patched by me)
+
+2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_sip.c: These flag tests were illogical. They were
+ testing sip_peer flags on a sip_pvt. Thanks to Russell for
+ helping to get this odd problem figured out.
+
+2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher at digium.com>
+
+ * apps/app_voicemail.c: When we reset the password via an external
+ command, we should also reset the password stored in the
+ in-memory list, too (otherwise it doesn't really take effect).
+ (closes issue #11809) Reported by: davetroy Patches:
+ fix_externpass.diff uploaded by davetroy (license 384)
+
+ * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
+ too
+
+ * main/acl.c: Just confirmed that all current platforms need this
+ header file
+
+2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Thanks to Russell's education I realize that
+ BUFSIZ has changed since I learned the C language over 20 years
+ ago... Resetting chan_sip to the size of BUFSIZ that I expected
+ in my old head to avoid to heavy memory allocations on some
+ systems.
+
+2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher at digium.com>
+
+ * main/acl.c: Fix the defines for OS X (and Solaris, too)
+
+2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
+ apps/app_followme.c: Add more dependencies on chan_local and add
+ a note to the description of chan_local so that people don't
+ disable it in menuselect just to clean up.
+
+ * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
+ still runs without chan_local, but will be missing forwarding
+ functionality.
+
+2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher at digium.com>
+
+ * main/acl.c: Ensure that we can get an address even when we don't
+ have a default route. (closes issue #9225) Reported by: junky
+ Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
+ (license 14) Tested by: oej, loloski, sergee
+
+2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej at edvina.net>
+
+ * channels/chan_sip.c: Cleaning up some documentation that led to
+ confusion in a bug report
+
+2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_local.c: Fixing an issue wherein monitoring local
+ channels was not possible. During a channel masquerade, the
+ monitors on the two channels involved are swapped. In 99% of the
+ cases this results in the desired effect. However, if monitoring
+ a local channel, this caused the monitor which was on the local
+ channel to get moved onto a channel which is immediately hung up
+ after the masquerade has completed. By swapping the monitors
+ prior to the masquerade, we avoid the problem by tricking the
+ masquerade into placing the monitor back onto the channel where
+ we want it. During the investigation of the issue, the channel's
+ monitor was the only thing that was swapped in such a manner
+ which did not make sense to have done. All other variable
+ swapping made sense.
+
+2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h: Permit the user to specify number of
+ seconds that a connection may remain idle, which fixes a crash on
+ reconnect with the MyODBC driver. (closes issue #11798) Reported
+ by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Bump the buffer size for Via headers up to
+ 512. There are some exceptionally large Via headers out there.
+ (closes issue #11783) Reported by: ofirroval
+
+2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell at digium.com>
+
+ * main/slinfactory.c: Fix a couple of memory leaks with frame
+ handling. Specifically, ast_frame_free() needed to be called on
+ the frame that came from the translator to signed linear.
+
+2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp at digium.com>
+
+ * include/asterisk/channel.h: Remove the __ in front of the unused
+ variable. This causes some compilers to freak out.
+
+2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell at digium.com>
+
+ * include/asterisk/translate.h, main/frame.c: Revert adding the
+ packed attribute, as it really doesn't make sense why that would
+ do any good. Fix the real bug, which is to do the check to see if
+ the frame came from a translator at the beginning of
+ ast_frame_free(), instead of at the end. This ensures that it
+ always gets checked, even if none of the parts of the frame are
+ malloc'd, and also ensures that we aren't looking at free'd
+ memory in the case that it is a malloc'd frame. (closes issue
+ #11792, reported by explidous, patched by me)
+
+ * include/asterisk/translate.h: Since we're relying on the offset
+ between the frame and the beginning of the translator pvt struct,
+ set the packed attribute to make sure we get to the right place.
+ (potential fix for issue #11792)
+
+2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson at digium.com>
+
+ * res/res_features.c: This should at least temporarily fix a
+ problem where the 't' Dial option is incorrectly passed to the
+ transferee when built-in attended transfers are used. There is
+ still a problem with 'T', but better to fix some problems than no
+ problems while we work on it. (closes issue #7904) Reported by:
+ k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
+ (license 138) Tested by: sergee, otherwiseguy
+
+2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand at digium.com>
+
+ * configs/cdr.conf.sample: doh! revert a revert of a revert
+ (changed by mistake in 99010)
+
+ * main/manager.c, configs/cdr.conf.sample: missed that one while
+ reverting
+
+ * main/manager.c: reverting 99001 - We need the Max-Age for
+ extending the life of cookie mansession_id
+
+2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell at digium.com>
+
+ * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
+ Have IAX2 optimize the codec translation path just like chan_sip
+ does it. If the caller's codec is in our codec list, move it to
+ the top to avoid transcoding. (closes issue #10500) Reported by:
+ stevedavies Patches: iax-prefer-current-codec.patch uploaded by
+ stevedavies (license 184) iax-prefer-current-codec.1.4.patch
+ uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
+ sheldonh
+
+2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming at digium.com>
+
+ * main/manager.c: we should only send the Set-Cookie header to the
+ browser on the first response after creating a manager session,
+ not on every response (doing so causes the browser to clear any
+ local cookies it may have associated with the session)
+
+2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker at digium.com>
+
+ * configs/zapata.conf.sample: Add a clarification about the
+ immediate= option of zapata.conf Issue 11784, patch by klaus3000.
+
+2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell at digium.com>
+
+ * .cleancount, include/asterisk/channel.h: Add an unused pointer to
+ the ast_channel struct. This makes the ast_channel structure
+ retain the same size as it had in previous 1.4 releases. Also,
+ all of the offsets for members in the structure are still the
+ same (except for the two pointers that got replaced for the new
+ spy/whisper architecture.)
+
+2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp at digium.com>
+
+ * .cleancount: Bump up cleancount due to previous commit that
+ changed the channel structure.
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
+ main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
+ (added), main/Makefile, include/asterisk/chanspy.h (removed),
+ include/asterisk/channel.h, main/audiohook.c (added): Replace
+ current spy architecture with backport of audiohooks. This should
+ take care of current known spy issues.
+
+ * channels/chan_iax2.c: Add missing NULLs at end of two
+ ast_load_realtimes. (closes issue #11769) Reported by: tequ
+ Patches: chaniax.patch uploaded by dimas (license 88)
+
+2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson at digium.com>
+
+ * channels/chan_local.c: Fix a deadlock in chan_local in
+ local_hangup. There was contention because the local_pvt was held
+ and it was attempting to lock a channel, which is the incorrect
+ locking order. (closes issue #11730) Reported by: UDI-Doug
+ Patches: 11730.patch uploaded by putnopvut (license 60) Tested
+ by: UDI-Doug
+
+2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp at digium.com>
+
+ * main/dial.c: Introduce a lock into the dialing API that protects
+ it when destroying the structure. (closes issue #11687) Reported
+ by: callguy Patches: 11687.diff uploaded by file (license 11)
+
+ * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
+ issue #11777) Reported by: tootai
+
+ * channels/chan_sip.c: Don't drop the old record route information
+ when dealing with packets related to a reinvite. (closes issue
+ #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
+ uploaded by kebl0155 (license 356)
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
+ versions use a separate library for some things so we need to use
+ it if present in codec_speex. (closes issue #11693) Reported by:
+ yzg
+
+2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell at digium.com>
+
+ * channels/chan_sip.c: Change a buffer in check_auth() to be a
+ thread local dynamically allocated buffer, instead of a massive
+ buffer on the stack. This fixes a crash reported by Qwell due to
+ running out of stack space when building with LOW_MEMORY defined.
+ On a very related note, the usage of BUFSIZ in various places in
+ chan_sip is arbitrary and careless. BUFSIZ is a system specific
+ define. On my machine, it is 8192, but by definition (according
+ to google) could be as small as 256. So, this buffer in
+ check_auth was 16 kB. We don't even support SIP messages larger
+ than 4 kB! Further usage of this define should be avoided, unless
+ it is used in the proper context.
+
+ * main/rtp.c, include/asterisk/translate.h, main/frame.c,
+ main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
+ codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
+ some memory access errors pointed out by the valgrind2.txt output
+ on issue #11698. The issue here is that it is possible for an
+ instance of a translator to get destroyed while the frame
+ allocated as a part of the translator is still being processed.
+ Specifically, this is possible anywhere between a call to
+ ast_read() and ast_frame_free(), which is _a lot_ of places in
+ the code. The reason this happens is that the channel might get
+ masqueraded during this time. During a masquerade, existing
+ translation paths get destroyed. So, this patch fixes the issue
+ in an API and ABI compatible way. (This one is for you,
+ paravoid!) It changes an int in ast_frame to be used as flag
+ bits. The 1 bit is still used to indicate that the frame contains
+ timing information. Also, a second flag has been added to
+ indicate that the frame came from a translator. When a frame with
+ this flag gets released and has this flag, a function is called
+ in translate.c to let it know that this frame is doing being
+ processed. At this point, the flag gets cleared. Also, if the
+ translator was requested to be destroyed while its internal frame
+ still had this flag set, its destruction has been deffered until
+ it finds out that the frame is no longer being processed.
+ Admittedly, this feels like a hack. But, it does fix the issue,
+ and I was not able to think of a better solution ...
+
+2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_sip.c: Based on the boundary found move over the
+ correct amount. (closes issue #11750) Reported by: tasker
+
+ * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
+ in the multipart mixed content type. (closes issue #11750)
+ Reported by: tasker
+
+2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
+ I added. Thanks to joetester on IRC for pointing this out.
+
+2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell at digium.com>
+
+ * main/translate.c: Revert a change that introduces an unacceptable
+ performance hit and is causing memory leaks ... (from rev 97973)
+
+2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Fixing another compilation error. I'm a bit off
+ today :(
+
+ * apps/app_queue.c: Oops. Last commit had compilation error.
+
+ * apps/app_queue.c: Adding explicit defaults for missing options to
+ init_queue. This is necessary because if a user either removes or
+ comments one of these options and reloads their queues, the
+ option will not reset to its default, instead maintaining the
+ value from prior to the reload. Thanks to John Bigelow for
+ pointing this error out to me.
+
+2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_odbc.c: Add a connection timeout attribute, as that was
+ what was intended with the login timeout, but ODBC divides it up
+ into 2 different timeouts. (Closes issue #11745)
+
+2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
+ (closes issue #11646) Reported by: caio1982 Patches:
+ dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
+ dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
+ by: caio1982, mvanbaak
+
+2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand at digium.com>
+
+ * main/http.c: Comment explaining how to force browser to always
+ read some html files from server.
+
+2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp at digium.com>
+
+ * main/rtp.c: If the incoming RTP stream changes codec force the
+ bridge to break if the other side does not support it. (closes
+ issue #11729) Reported by: tsearle Patches:
+ new_codec_patch_udiff.patch uploaded by tsearle (license 373)
+
+ * res/res_agi.c: If the channel is hungup during RECORD FILE send a
+ result code of -1 to be uniform with everything else. (closes
+ issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
+ by davevg (license 209)
+
+2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson at digium.com>
+
+ * main/channel.c: Properly report the hangup cause as no answer
+ when someone does not answer (closes issue #10574, reported by
+ boch, patched by moy)
+
+2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher at digium.com>
+
+ * codecs/gsm/Makefile: Add another exception (which doesn't work)
+ for -march optimization flag. Reported by: thomasmebes Patch by:
+ tilghman (Closes issue #11563)
+
+2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell at digium.com>
+
+ * doc/security.txt, main/asterisk.c, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ makeopts.in: Backport the ability to set the ToS bits on Linux
+ when not running as root. Normally, we would not backport
+ features into 1.4, but, I was convinced by the justification
+ supplied by the supplier of this patch. He pointed out that this
+ patch removes a requirement for running as root, thus reducing
+ the potential impacts of security issues. (closes issue #11742)
+ Reported by: paravoid Patches: libcap.diff uploaded by paravoid
+ (license 200)
+
+2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_followme.c: Ensure the return value of ast_bridge_call
+ is passed back up as the application return value. This is needed
+ for transfers to function so the PBX core knows to continue
+ execution. (closes issue #10327) Reported by: kkiely
+
+2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: Back out changes from revision 97077, since
+ it wasn't perfect
+
+2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell at digium.com>
+
+ * main/frame.c: Fix samples vs. length calculations for g722
+
+ * main/translate.c: Simplify this code with a suggestion from Luigi
+ on the asterisk-dev list. Instead of using is16kHz(), implement a
+ format_rate() function.
+
+ * main/translate.c: Fix various timing calculations that made
+ assumptions that the audio being processed was at a sample rate
+ of 8 kHz.
+
+2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c, main/translate.c: 1) When we get a
+ translated frame out, clone it, because if the translator pvt is
+ freed before we use the frame, bad things happen. 2) Getting a
+ failure from ast_sched_delete means that the schedule ID is
+ currently running. Don't just ignore it. (Closes issue #11698)
+
+2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
+ have logged into VoiceMailMain and chosen to leave a message.
+ (closes issue #11735, reported and patched by jamessan)
+
+2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf at digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
+ same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
+ overrode the normally generate yyfree func with our own version
+ that checks the pointer for non-null before passing to free().
+ Also takes care of a little problem with 2.5.33 and the use of
+ the __STDC_VERSION__ macro.
+
+ * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
+ fix for 2 things: a problem Terry was having in OSX with null
+ pointers, which was my fault, as I probably forgot to run the sed
+ script last time I made mods. So, I moved the fix into the flex
+ input itself. Then, I found when I used flex 2.5.33, that it was
+ using __STDC_VERSION__, and that's not real good; so I added back
+ in a DIFFERENT sed script to fix that little mess. Tested
+ everything, a couple different ways. Hope I did no harm, at the
+ least.
+
+2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker at digium.com>
+
+ * include/asterisk/frame.h: Fix a comment that is no longer true.
+
+2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
+ pbx/kdeconsole_main.cc (removed): Remove other remnants of
+ pbx_kdeconsole
+
+ * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
+ worked in ages, and nobody has complained. (closes issue #11706,
+ reported by caio1982)
+
+2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp at digium.com>
+
+ * funcs/func_groupcount.c: Don't try to copy the category from the
+ group if no category exists. (closes issue #11724) Reported by:
+ IgorG Patches: group_count.v1.patch uploaded by IgorG (license
+ 20)
+
+2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
+ messages
+
+ * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
+ actually load on my system still (related to issue #11706)
+
+2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker at digium.com>
+
+ * main/cli.c: Correctly display a message if a command could not be
+ found. Also fix a comment which may have led to this happening.
+ Issue 11718, reported by kshumard.
+
+ * main/cli.c: Fix some locking and return value funkiness. We
+ really shouldn't be unlocking this lock inside of a function,
+ unless we locked it there too.
+
+2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
+ smaller functions this time
+
+2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell at digium.com>
+
+ * res/res_features.c: Fix saying the parking space number to the
+ caller doing the parking ...
+
+2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming at digium.com>
+
+ * codecs/codec_zap.c: report the same message whether Zaptel does
+ not have transcoder support loaded or no transcoders were found
+
+2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
+ function. As underlined in issue #10437 by Josh, we need to
+ prevent a possible memory leak. We only set the name part of the
+ caller id, the number part is not relevant when dealing with
+ JIDs. Closes issue #11549.
+
+2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp at digium.com>
+
+ * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
+ Monitor is running on the channel. (closes issue #11709) Reported
+ by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
+ (license 371)
+
+2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming at digium.com>
+
+ * channels/chan_zap.c: pass the right variable to get an error
+ string... oops
+
+ * channels/chan_zap.c: add error number output to ioctl failure
+ messages to help with debugging
+
+2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher at digium.com>
+
+ * main/cli.c, main/editline/readline.c: Allow filename completion
+ on zero-length modules, remove a memory leak, remove a file
+ descriptor leak, and make filename completion thread-safe.
+ Patched and tested by tilghman. (Closes issue #11681)
+
+2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: use the \retval doxygen command properly
+
+ * apps/app_queue.c: Part 1 of N of adding doxygen comments to
+ app_queue. I picked some of the most common functions used (which
+ also happen to be some the biggest/ugliest functions too) to
+ document first. I'm pretty new to doxygen so criticism is
+ welcome.
+
+ * apps/app_queue.c: Some coding guidelines-related cleanup
+
+2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp at digium.com>
+
+ * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
+ (closes issue #11443) Reported by: eferro Patches:
+ dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
+ 337)
+
+2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher at digium.com>
+
+ * main/autoservice.c, main/utils.c: Increase constants to where
+ we're less likely to hit them while debugging. (Closes issue
+ #11694)
+
+2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_voicemail.c: Making some changes designed to not allow
+ for a corrupted mailstream for a vm_state. 1. Add locking to the
+ vm_state retrieval functions so that no linked list corruption
+ occurs. 2. Make sure to always grab the persistent vm_state when
+ mailstream access is necessary. 3. Correct an incorrect return
+ value in the init_mailstream function. (closes issue #11304,
+ reported by dwhite)
+
+2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp at digium.com>
+
+ * funcs/func_groupcount.c: If no group has been provided to the
+ GROUP_COUNT dialplan function then use the first one specific to
+ the channel. (closes issue #11077) Reported by: m4him
+
+ * apps/app_queue.c: Make app_queue calls work with directed pickup.
+ (closes issue #11700) Reported by: jbauer
+
+2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher at digium.com>
+
+ * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
+ found in issue #11386, but not completely closing that issue.
+
+2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell at digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 96931 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
+ 2 lines Change misery.digium.com to pbx.digium.com ........
+
+ * res/res_smdi.c: Don't crash if something happens when setting up
+ an SMDI interface and it gets destroyed before the SMDI port
+ handling thread gets created.
+
+2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan at gmail.com>
+
+ * res/res_jabber.c: Indentation fix, makes the code easier to read
+
+ * res/res_jabber.c: Compute the base64 value over the
+ [authzid]\0authcid\0password string, thus excluding the trailing
+ NULL byte. This change has already been committed to trunk, see
+ #11644.
+
+2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell at digium.com>
+
+ * main/devicestate.c: Don't pass an empty string as the device
+ name.
+
+2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher at digium.com>
+
+ * main/devicestate.c: Fix the problem of notification of a device
+ state change to a device with a '-' in the name. Could probably
+ do with a better fix in trunk, but this bug has been open way too
+ long without a better solution. Reported by: stevedavies Patch
+ by: tilghman (Closes issue #9668)
+
+2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker at digium.com>
+
+ * res/res_features.c: Properly continue in the dialplan if using
+ PARKINGEXTEN and the slot is full. Issue 11237, patch by me.
+
+2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher at digium.com>
+
+ * channels/chan_sip.c: If you change the bindaddr in sip.conf to a
+ non-bound address and reload, sip goes kablooie. Reported and
+ patched by: one47 (Closes issue #11535)
+
+2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell at digium.com>
+
+ * channels/chan_zap.c: Make use of the temporary channel pointer
+ while the pvt is unlocked. (closes issue #11675) Reported by:
+ flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
+ uploaded by flefoll (license 244)
+
+ * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
+ been destroyed before we get to this point (closes issue #11672,
+ reported by snuffy, patched by me)
+
+2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher at digium.com>
+
+ * res/res_config_pgsql.c: Missed initialization caused crash.
+ Reported and fixed by: tiziano (Closes issue #11671)
+
+2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter at beronet.com>
+
+ * channels/chan_misdn.c: make sure frame is completely clean,
+ before we send it to asterisk as DTMF. If we don't make it clean,
+ it happens that one way audio occurs..
+
+ * channels/chan_misdn.c: when overlapdial was used and no number
+ was dialed, the call was dropped, now we just jump into the s
+ extension, which makes a lot more sense.
+
+2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson at digium.com>
+
+ * apps/app_queue.c: We need to reset the membername to NULL on each
+ iteration of this loop, otherwise the result is that multiple
+ members can have the same name, since the variable was not reset
+ on each iteration of the loop.
+
+2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell at digium.com>
+
+ * pbx/pbx_config.c: Convert locks of the contexts list in
+ pbx_config to the appropriate rdlock or wrlock
+
+ * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
+ list.
+
+ * apps/app_macro.c: app_macro only needs a rdlock on the contexts
+ list.
+
+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
+
[... 14469 lines stripped ...]
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