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are in separate files (still #include'd because of tangling in the data
structures, but this is going to be cleaned up).
The video grabbing functions still need to be moved to a separate file.
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character. Also, fix the documentation to match the code.
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Remove an unused keypad field and some debugging messages.
Adjust formatting on config file parsing
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With this change we can do
setenv SDL_VIDEODRIVER aalib
and output to an ascii window (which is still in an X11 window).
If you also do
unsetenv DISPLAY
then the output goes into the main asterisk window, unfortunately
it interferes with the normal output so you don't see much.
In any case, i don't think we are very far away from having a working
xterm videophone!
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do not change.
Ths masks (but does not solve) a but that i am seeing in doing a
'gmake install' without donig a 'gmake all' first.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007) | 2 lines
Argh... I suppose third time's the charm.
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from 'keypad_entry' to 'region'. Fix the example file accordingly.
Also make some fixes in the code do reset entries on reload of the keypad.
The recently committed kpad2.jpg has the correct names.
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console_video.c
This will ease the task of splitting console_video.c into its components
(V4L and X11 grabbers, various video codecs and packetizers, SDL),
as well as ease future extensions (e.g. additional video sources,
codecs and rendering engines).
For the time being nothing changes for users: video support is off by
default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included
(if SDL and FFMPEG are available).
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parameters, as well as RLT support for DMS type switches
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94256 | russell | 2007-12-20 14:22:22 -0600 (Thu, 20 Dec 2007) | 13 lines
Merged revisions 94255 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) | 10 lines
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
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see description in config.h .
They are a variant of the set of macros i used in chan_oss.c,
structured in a way to be more robust to the presence of
spurious ';' - basically, they define wrappers for 'do {'
and '} while (0)', plus some helper functions to deal with simple
cases such as ast_copy_string, ast_malloc, strtoul, ast_true ...
The prefix (CV_ as 'Config Variable') tries to be easy to remember
and has been chosen to not conflict with other existing macros in the tree.
For the time being, I have only updated the three source files in the
tree that used the old M_* macros. Hopefully, more files will be
converted.
NOTE:
I understand that inventing my own dialect of C is generally wrong;
however, the lack of adequate support in the language encourages
lazy programming practices (such as ignoring errors, bounds, etc.)
and this increases the chance of vulnerability in the code, especially
because we are parsing user input here.
Hopefully, these macros and the use of ast_parse_arg (in config.h)
should encourage the programmer to write more robust code.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007) | 3 lines
Avoid segfault in chan_iax when peer isn't defined
(Closes issue #11602)
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by adding an exclamation mark to the dial string.
This patch also exists for 1.4 in the fixtoheader-1.4 branch
and has been in production for quite some time.
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string as a NAT keepalive packet.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r93668 | tilghman | 2007-12-18 12:29:39 -0600 (Tue, 18 Dec 2007) | 10 lines
Merged revisions 93667 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines
Fixing AST-2007-027 (Closes issue #11119)
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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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At this point WINARCH should go away.
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provided you look for soundcard.h in the right place...
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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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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
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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.
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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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- 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)
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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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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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Reported by: snuffy
Patch by: snuffy,tilghman
(Closes issue #11315)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 lines
Up the length of the format on the SIP channel since it can now be rather long.
(closes issue #11552)
Reported by: francesco_r
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r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 Dez 2007) | 1 line
fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread starting.
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Reported by: snuffy
Patch by: snuffy
(Closes issue #11547)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) | 5 lines
Properly initialize polarity statuses, so that they are detected properly.
Reported by: julianjm
Patch by: julianjm
(Closes issue #10238)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10690)
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r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines
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.
Issue 10690.
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reported
issue #11449 has demonstrated that it actually was a performance hit on his
machine. I think that it is possible that it could still be a benefit on systems
under higher load, especially SMP systems, but I don't have enough time or interest
to find out at the moment.
(closes issue #11449)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11048)
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r92617 | qwell | 2007-12-12 15:15:45 -0600 (Wed, 12 Dec 2007) | 4 lines
Don't increment user count until after name has been recorded (if enabled).
Issue 11048, tested by pep.
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misdn stuff needs a lot of doxygenification
(Hello, Qwell :-) )
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structures missing. Patched configure to check for this stuff and
put a #ifdef around the offending code in chan_zap. Thanks to file
for overseeing this.
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the match..
Closes issue #11518.
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