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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines
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)
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modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
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There are some left in the .flex files, left to the maintainer...
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globally visible.
Document the fact that DEFAULT_TMP_DIR cannot be overridden
from the default configuration (this needs to be fixed, as you
could have a totally different spooldir configured at runtime,
and yet DEFAULT_TMP_DIR keeps the compile-time default).
Remove two unused entries for sounds and images.
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words in MODULE_PREFIX.
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+ document the difference between "A=foo make ..." and "make A=foo ..."
and suggest using COPTS/LDOPTS if you need to use the second form to
pass compiler and loader flags;
+ define only in one place the environment used to build stuff in menuselect/
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At this point WINARCH should go away.
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r93420 | qwell | 2007-12-17 16:56:58 -0600 (Mon, 17 Dec 2007) | 1 line
Missed a spot..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 lines
What was I thinking when I wrote this masterpiece?
-1 + 1 = 0.. who woulda thunk it?.
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provided you look for soundcard.h in the right place...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 lines
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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17 Dec 2007) | 6 lines
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. ;-)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines
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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec 2007) | 6 lines
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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6 lines
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)
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and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please)
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can not be accessed directly in the SIP pvt structure. All RTP related timeouts have to be retrieved using the ast_rtp_* function calls.
(closes issue #11562)
Reported by: ibc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11576)
Reported by: junky
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you want to pthread_join you have to explicitly create it as joinable, and also if using pthread_cancel you have to have a pthread_testcancel to see if it has been called.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r93183 | kpfleming | 2007-12-16 23:21:08 -0800 (Sun, 16 Dec 2007) | 2 lines
fix some copy-and-paste leftovers
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines
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.
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(closes issue #11572)
Reported by: dimas
Patches:
dundilog-trunk.patch uploaded by dimas (license 88)
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you need to reload after changes. Thanks YS.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(related to issue #11414)
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for a
manager realtime implementation.
If you change accounts in manager.conf, you now need to reload to activate the
changes (deletions, additions). This was not the case with 1.4.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(closes issue #11414)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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date,
even with experimental stuff.
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: slavon
Patches:
asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich (license 176)
Tested by: eliel
(closes issue #9325)
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non-responsive SIP servers.
Thanks, jcmoore, for the patch!
Reported by: jcmoore
Patches:
peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9)
(closes issue #9771)
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Didn't a clever guy once say "Compile before you commit" ?
:-)
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This is disabled in the default build, you need to explicitly enable it
compiling with
make COPTS=-DHAVE_VIDEO_CONSOLE
In return, you will be able to do a video call with chan_oss, using
the webcam (or X11 grabbing) as local source, and rendering the
incoming stream on your screen. Currently supported formats are
h261, h263, h263+, h264, mpeg4 (all through the avcodec lib, part
of ffmpeg).
Incoming video is on the left, outgoing video is on the right,
while the center displays a keypad (if configured so).
Right clicking on the video windows increases the size,
center clicking reduces the size.
Dragging the mouse (with the left key) on the right window
while the X11 grabber is active moves the grab area.
This is the result of work by Sergio Fadda, Marta Carbone
and myself, all properly disclaimed to digium.
Note, there is a lot of work left to do in this module,
including adding support for Video4LinuxV2 (I have patches
from Matteo Brancaleoni which should be integrated),
and making the GUI a lot more friendly than it is now
(e.g. supporting merging or switching among multiple sources,
a text window, and more).
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setting from voicemail.conf instead. This commit will help to get issue
#11415 moving towards commitment.
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Reported by: snuffy
Patch by: snuffy,tilghman
(Closes issue #11315)
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