1) Add the Dialplan class, for NewExten and VarSet events, which should cut
down on the volume of traffic in the Call class.
2) Permit some commands to be run from multiple classes, such as allowing
DBGet to be run from either the System or the Reporting class.
3) Heavily document each class in the sample config, as there were several
that made no sense to be in the write= line, and two that made no sense to be
in the read= line (since they controlled no permissions there).
(Closes issue #10386)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- support scrolling of message window;
- simplify the code for creating a message window,
and try it using a second one in the top of
the keypad (where we echo the dialed number).
The 'skin' that supports these two windows will be
committed separately.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines
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.
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a number to dial in the 'message' area under the
keypad.
Now you can make calls using the keypad as a regular phone
(or the keyboard for chars not present on the keypad)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
commands so you can start and stop the gui even outside
of a call. This is convenient for testing, and also for
using the keypad to pick up a call, and to dial a number
(the latter not yet implemented, but should be close).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines
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)
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remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
of a hack. It just asks the core to generate the same tone that it would when
you hear ringback when making an outbound call. But hey, it works, and you get
the localized ring tone for the appropriate language set on the channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working)
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to an 8 kHz endpoint, then codec_resample will automatically be used to properly
resample the audio before sending it to/from chan_console.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface. It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.
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to add more entries. This required moving struct grab_desc to the common
header, and adding an entry in the Makefile.
On passing, cleanup some comments and file headers (some are still missing).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines
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
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r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SDL is also detected at runtime).
Now we should be able to stream video even without a rendering device
(useful for remote monitoring).
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in the peers container after a reload. Somehow, this bug doesn't exist in 1.4 ...
(closes issue #11626)
(reported by pnlarsson, additional info from mvanbaak, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Tilghman Lesher
Luigi Rizzo
Philippe Sultan
Kevin P. Fleming
Olle Johansson
Joshua Colp
Russell Bryant
Christian Richter
Mark Michelson
Matthew Fredrickson