[asterisk-commits] murf: branch murf/utf8-whatif r112519 - in /team/murf/utf8-whatif: ./ apps/ b...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Apr 2 14:24:00 CDT 2008
Author: murf
Date: Wed Apr 2 14:23:58 2008
New Revision: 112519
URL: http://svn.digium.com/view/asterisk?view=rev&rev=112519
Log:
Merged revisions 96706-112466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r96715 | russell | 2008-01-05 14:35:54 -0700 (Sat, 05 Jan 2008) | 3 lines
Add a note about changing modules.conf since another console channel driver is
now present that can not be used at the same time as chan_alsa or chan_oss.
................
r96716 | russell | 2008-01-05 15:04:08 -0700 (Sat, 05 Jan 2008) | 3 lines
Print out the name of a function being registered in color, just like the name
of applications when they get registered.
................
r96717 | russell | 2008-01-05 15:09:06 -0700 (Sat, 05 Jan 2008) | 15 lines
Now that the version.h file was getting properly regenerated every time the svn
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.
................
r96743 | russell | 2008-01-05 16:05:35 -0700 (Sat, 05 Jan 2008) | 4 lines
Convert this file over the new method of getting the Asterisk version.
(I don't have this building on this machine, so caio1982 on IRC is going to
test it for me. :) )
................
r96776 | rizzo | 2008-01-07 04:22:01 -0700 (Mon, 07 Jan 2008) | 4 lines
resolve a load-time problem avoiding a call to console_do_answer.
On passing, fix dialling from the keypad.
................
r96836 | rizzo | 2008-01-07 08:06:54 -0700 (Mon, 07 Jan 2008) | 4 lines
update comments to reflect reality (or at least planned behaviour).
minor code cleanups
................
r96858 | file | 2008-01-07 08:52:55 -0700 (Mon, 07 Jan 2008) | 6 lines
Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. Previously they would get registered twice because of the way manager.c operates.
(closes issue #11699)
Reported by: caio1982
Patches:
manager_module_commands1.diff uploaded by caio1982 (license 22)
................
r96862 | kpfleming | 2008-01-07 09:17:31 -0700 (Mon, 07 Jan 2008) | 2 lines
add a file-format driver for 16KHz signed linear... which may or may not work
................
r96885 | russell | 2008-01-07 09:42:42 -0700 (Mon, 07 Jan 2008) | 11 lines
Merged revisions 96884 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) | 3 lines
Don't crash if something happens when setting up an SMDI interface and it gets
destroyed before the SMDI port handling thread gets created.
........
................
r96886 | russell | 2008-01-07 09:59:04 -0700 (Mon, 07 Jan 2008) | 2 lines
Build the HTML version of the doc files for tarballs, as well
................
r96887 | russell | 2008-01-07 10:09:23 -0700 (Mon, 07 Jan 2008) | 3 lines
If the HTML documentation exists, install it in the static-http/docs directory
so that it can be viewed through the Asterisk http server if it is turned on.
................
r96888 | russell | 2008-01-07 10:15:11 -0700 (Mon, 07 Jan 2008) | 2 lines
Add a note about viewing the default set of documentation using the built-in http server
................
r96933 | russell | 2008-01-07 13:48:23 -0700 (Mon, 07 Jan 2008) | 18 lines
Merged revisions 96932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r96932 | russell | 2008-01-07 14:47:52 -0600 (Mon, 07 Jan 2008) | 10 lines
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
........
................
................
r96934 | mmichelson | 2008-01-07 14:04:09 -0700 (Mon, 07 Jan 2008) | 7 lines
Adding user-configurable TCP timeout settings to IMAP voicemail. This could
go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.
(closes issue #11665, reported by yehavi)
................
r96935 | mmichelson | 2008-01-07 14:10:59 -0700 (Mon, 07 Jan 2008) | 4 lines
Document some weird casting magic that's necessary to interface
with the c-client
................
r96936 | qwell | 2008-01-07 14:12:33 -0700 (Mon, 07 Jan 2008) | 4 lines
Display a message if no config mappings are found with "core show config mappings".
Closes issue #11704, patch by kshumard.
................
r96987 | mmichelson | 2008-01-07 15:31:03 -0700 (Mon, 07 Jan 2008) | 3 lines
Explicitly make literal constants long where they are expected to be.
................
r96988 | rizzo | 2008-01-07 16:03:11 -0700 (Mon, 07 Jan 2008) | 5 lines
add support for cropping the keypad image while displaying it.
This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
................
r97041 | file | 2008-01-08 08:26:50 -0700 (Tue, 08 Jan 2008) | 4 lines
Make ast_atomic_fetchadd_int_slow magically appear in extconf.
(closes issue #11703)
Reported by: dmartin
................
r97103 | file | 2008-01-08 11:42:39 -0700 (Tue, 08 Jan 2008) | 12 lines
Merged revisions 97093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4 lines
Make app_queue calls work with directed pickup.
(closes issue #11700)
Reported by: jbauer
........
................
r97125 | tilghman | 2008-01-08 12:06:27 -0700 (Tue, 08 Jan 2008) | 11 lines
Merged revisions 97077 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines
Apply multiple crash fixes, found in issue #11386, but not completely
closing that issue.
........
................
r97153 | file | 2008-01-08 12:55:56 -0700 (Tue, 08 Jan 2008) | 12 lines
Merged revisions 97152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan 2008) | 4 lines
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
........
................
r97154 | file | 2008-01-08 13:06:52 -0700 (Tue, 08 Jan 2008) | 6 lines
Move common code for setting T38 capabilities and fix a bug with fax detection in the SIP RTP read callback. It's still sort of silly... but more on that later.
(closes issue #11239)
Reported by: dimas
Patches:
sipt38prop.patch uploaded by dimas (license 88)
................
r97193 | mmichelson | 2008-01-08 13:45:59 -0700 (Tue, 08 Jan 2008) | 17 lines
Merged revisions 97192 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines
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)
........
................
r97196 | file | 2008-01-08 13:50:57 -0700 (Tue, 08 Jan 2008) | 14 lines
Merged revisions 97195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 lines
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)
........
................
r97197 | file | 2008-01-08 13:52:07 -0700 (Tue, 08 Jan 2008) | 2 lines
One line documentation ftw!
................
r97198 | tilghman | 2008-01-08 13:56:38 -0700 (Tue, 08 Jan 2008) | 11 lines
Merged revisions 97194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines
Increase constants to where we're less likely to hit them while debugging.
(Closes issue #11694)
........
................
r97199 | oej | 2008-01-08 13:58:56 -0700 (Tue, 08 Jan 2008) | 2 lines
Iksemel is alive in a new home. Release 1.3 is out with bug fixes.
................
r97200 | oej | 2008-01-08 14:01:46 -0700 (Tue, 08 Jan 2008) | 4 lines
Change reference to external library so it appears on the extref listing
http://www.asterisk.org/doxygen/trunk/extref.html
................
r97203 | mmichelson | 2008-01-08 14:18:32 -0700 (Tue, 08 Jan 2008) | 8 lines
Adding the option of specifying a second interface in a member definition for a queue. app_queue
will monitor this second device's state for the member, even though it actually calls the first
interface. This ability has been added for statically defined queue members, realtime queue members,
and dynamic queue members added through the CLI, dialplan, or manager.
(closes issue #11603, reported by acidv)
................
r97208 | mmichelson | 2008-01-08 14:28:04 -0700 (Tue, 08 Jan 2008) | 10 lines
Blocked revisions 97206 via svnmerge
........
r97206 | mmichelson | 2008-01-08 15:24:48 -0600 (Tue, 08 Jan 2008) | 3 lines
Some coding guidelines-related cleanup
........
................
r97248 | twilson | 2008-01-08 14:56:57 -0700 (Tue, 08 Jan 2008) | 2 lines
Initialize new variable to NULL
................
r97280 | rizzo | 2008-01-08 16:09:44 -0700 (Tue, 08 Jan 2008) | 11 lines
add support for textareas, used for various dialog windows on the gui.
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.
................
r97281 | rizzo | 2008-01-08 16:11:25 -0700 (Tue, 08 Jan 2008) | 5 lines
add a font suitable for use with the console GUI.
The background of this particular image is transparent so we
can preserve the original background when we draw strings.
................
r97282 | rizzo | 2008-01-08 16:14:48 -0700 (Tue, 08 Jan 2008) | 6 lines
Add the annotation for the textarea used for messages,
and also change the background from white to something
different to show that we can make use of fonts with transparent
background.
................
r97303 | rizzo | 2008-01-08 16:35:05 -0700 (Tue, 08 Jan 2008) | 4 lines
add copyright (most of this code was written by Marta Carbone),
remove some unused code, add/clarify some comments.
................
r97305 | tilghman | 2008-01-08 16:51:51 -0700 (Tue, 08 Jan 2008) | 6 lines
Add a new flag 'd' (with optional context) permitting any extension within
that context to be entered as a new extension during the playback of a
voicemail greeting.
Patch inspired by bluecrow76, by tilghman.
(Closes issue #7063)
................
r97307 | mmichelson | 2008-01-08 16:56:24 -0700 (Tue, 08 Jan 2008) | 13 lines
Merged revisions 97304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan 2008) | 5 lines
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.
........
................
r97309 | mmichelson | 2008-01-08 17:18:15 -0700 (Tue, 08 Jan 2008) | 11 lines
Merged revisions 97308 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan 2008) | 3 lines
use the \retval doxygen command properly
........
................
r97364 | tilghman | 2008-01-08 17:51:59 -0700 (Tue, 08 Jan 2008) | 13 lines
Merged revisions 97350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) | 5 lines
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)
........
................
r97365 | tilghman | 2008-01-08 17:58:22 -0700 (Tue, 08 Jan 2008) | 2 lines
New option in trunk, needs strdupa to be safe, too
................
r97389 | rizzo | 2008-01-09 05:13:32 -0700 (Wed, 09 Jan 2008) | 4 lines
make get_video_desc() return the active console if
passed a null argument (channel).
................
r97390 | rizzo | 2008-01-09 05:23:18 -0700 (Wed, 09 Jan 2008) | 7 lines
implement the "console startgui" and "console stopgui"
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).
................
r97421 | kpfleming | 2008-01-09 08:31:56 -0700 (Wed, 09 Jan 2008) | 10 lines
Merged revisions 97410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) | 2 lines
add error number output to ioctl failure messages to help with debugging
........
................
r97449 | kpfleming | 2008-01-09 08:45:34 -0700 (Wed, 09 Jan 2008) | 10 lines
Merged revisions 97448 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) | 2 lines
pass the right variable to get an error string... oops
........
................
r97451 | file | 2008-01-09 09:13:24 -0700 (Wed, 09 Jan 2008) | 14 lines
Merged revisions 97450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6 lines
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)
........
................
r97488 | rizzo | 2008-01-09 09:44:20 -0700 (Wed, 09 Jan 2008) | 8 lines
Implement keyboard handling, and use it to enter
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)
................
r97490 | phsultan | 2008-01-09 09:59:09 -0700 (Wed, 09 Jan 2008) | 15 lines
Merged revisions 97489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
........
................
r97495 | kpfleming | 2008-01-09 10:30:13 -0700 (Wed, 09 Jan 2008) | 10 lines
Merged revisions 97491 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) | 2 lines
report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found
........
................
r97530 | rizzo | 2008-01-09 11:03:40 -0700 (Wed, 09 Jan 2008) | 11 lines
Two changes:
- 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.
................
r97531 | russell | 2008-01-09 11:04:14 -0700 (Wed, 09 Jan 2008) | 10 lines
Merged revisions 97529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | 2 lines
Fix saying the parking space number to the caller doing the parking ...
........
................
r97532 | rizzo | 2008-01-09 11:04:17 -0700 (Wed, 09 Jan 2008) | 4 lines
add annotations for the two message windows
we use.
................
r97533 | rizzo | 2008-01-09 11:12:44 -0700 (Wed, 09 Jan 2008) | 3 lines
remove a wrong 'const'
................
r97577 | mmichelson | 2008-01-09 11:53:29 -0700 (Wed, 09 Jan 2008) | 11 lines
Merged revisions 97575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan 2008) | 3 lines
Part 2 of app_queue doxygen improvements. Some smaller functions this time
........
................
r97620 | qwell | 2008-01-09 13:13:14 -0700 (Wed, 09 Jan 2008) | 9 lines
Merged revisions 97618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line
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.
........
................
r97623 | qwell | 2008-01-09 13:30:54 -0700 (Wed, 09 Jan 2008) | 14 lines
Merged revisions 97622 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11718)
........
r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines
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.
........
................
r97634 | twilson | 2008-01-09 14:37:26 -0700 (Wed, 09 Jan 2008) | 10 lines
Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
................
r97641 | russell | 2008-01-09 15:27:50 -0700 (Wed, 09 Jan 2008) | 11 lines
Merged revisions 97640 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | 3 lines
Make pbx_gtkconsole build ... but doesn't actually load on my system still
(related to issue #11706)
........
................
r97643 | russell | 2008-01-09 15:51:15 -0700 (Wed, 09 Jan 2008) | 3 lines
re-add check for gtk1, which is used for pbx_gtkconsole
(related to issue #11706)
................
r97644 | russell | 2008-01-09 15:53:12 -0700 (Wed, 09 Jan 2008) | 2 lines
re-gen configure
................
r97646 | russell | 2008-01-09 16:02:19 -0700 (Wed, 09 Jan 2008) | 10 lines
Merged revisions 97645 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | 2 lines
Strip terminal sequences from the verbose messages
........
................
r97650 | russell | 2008-01-09 17:11:02 -0700 (Wed, 09 Jan 2008) | 2 lines
Ensure that libg722.a gets rebuilt if one of the files changes
................
r97651 | tilghman | 2008-01-09 17:12:35 -0700 (Wed, 09 Jan 2008) | 11 lines
Several manager changes:
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)
................
r97652 | russell | 2008-01-09 17:17:02 -0700 (Wed, 09 Jan 2008) | 2 lines
Fix this so it doesn't force codec_g722 to get relinked every time
................
r97653 | twilson | 2008-01-09 17:22:50 -0700 (Wed, 09 Jan 2008) | 2 lines
Attempt at making lookup_iface work under FreeBSD. Not yet tested, but it compiles under OS X. And still works under linux.
................
r97655 | tilghman | 2008-01-09 17:39:34 -0700 (Wed, 09 Jan 2008) | 2 lines
oops, missed the case of a 0 permission (which should mean everybody is allowed, not nobody)
................
r97656 | murf | 2008-01-09 17:50:39 -0700 (Wed, 09 Jan 2008) | 1 line
The fixes in this commit are mainly to allow compiling of trunk with --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know.
................
r97657 | russell | 2008-01-09 17:54:01 -0700 (Wed, 09 Jan 2008) | 3 lines
These prototypes are not supposed to be in asterisk.h. They are already in
version.h.
................
r97698 | file | 2008-01-10 08:12:08 -0700 (Thu, 10 Jan 2008) | 14 lines
Merged revisions 97697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan 2008) | 6 lines
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)
........
................
r97745 | russell | 2008-01-10 09:15:43 -0700 (Thu, 10 Jan 2008) | 12 lines
Merged revisions 97734 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | 4 lines
Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has
complained.
(closes issue #11706, reported by caio1982)
........
................
r97758 | russell | 2008-01-10 09:22:10 -0700 (Thu, 10 Jan 2008) | 10 lines
Merged revisions 97753 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines
Remove other remnants of pbx_kdeconsole
........
................
r97767 | russell | 2008-01-10 09:27:03 -0700 (Thu, 10 Jan 2008) | 5 lines
Use AST_EXT_TOOL_CHECK() for the GTK check again. I changed this to an inline
implementation to fix a small bug, but after a discussion with rizzo, I went to
change it back. Also, it turns out that the implementation of the macro already
supported what was needed to fix the problem.
................
r97769 | russell | 2008-01-10 09:43:31 -0700 (Thu, 10 Jan 2008) | 2 lines
spaces to tabs
................
r97804 | russell | 2008-01-10 10:30:24 -0700 (Thu, 10 Jan 2008) | 2 lines
minor formatting changes
................
r97805 | tilghman | 2008-01-10 10:38:36 -0700 (Thu, 10 Jan 2008) | 3 lines
Fix problem with timestr going out of scope
(Closes issue #11726, closes issue #11731)
................
r97825 | twilson | 2008-01-10 12:03:04 -0700 (Thu, 10 Jan 2008) | 2 lines
Check pointers before freeing (was getting WARNINGS under MALLOC_DEBUG)
................
r97826 | twilson | 2008-01-10 12:07:36 -0700 (Thu, 10 Jan 2008) | 2 lines
heh, remove patch to generated file.
................
r97846 | mmichelson | 2008-01-10 13:05:43 -0700 (Thu, 10 Jan 2008) | 5 lines
Use the appropriate line ending for the X-Asterisk-VM-Message-Type header.
(closes issue #11734, reported and patched by jaroth)
................
r97848 | qwell | 2008-01-10 13:13:24 -0700 (Thu, 10 Jan 2008) | 9 lines
Merged revisions 97847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line
Fix a comment that is no longer true.
........
................
r97850 | murf | 2008-01-10 13:45:05 -0700 (Thu, 10 Jan 2008) | 9 lines
Merged revisions 97849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line
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.
........
................
r97890 | murf | 2008-01-10 14:46:56 -0700 (Thu, 10 Jan 2008) | 9 lines
Merged revisions 97889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line
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.
........
................
r97933 | mmichelson | 2008-01-10 14:58:47 -0700 (Thu, 10 Jan 2008) | 14 lines
Merged revisions 97925 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines
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)
........
................
r97974 | russell | 2008-01-10 16:10:00 -0700 (Thu, 10 Jan 2008) | 3 lines
Fix the buffer_samples value. For signed linear, the number of samples needed
to fill the buffer is half the buffer size.
................
r97975 | russell | 2008-01-10 16:16:09 -0700 (Thu, 10 Jan 2008) | 7 lines
Fix various issues in codec_g722.
- The most common fix being made here is to fix all of the places where the
number of output samples and output bytes gets updated in the translator
state structure.
- Fix a number of other places where the number of samples provided as an
initialization value to a struct was incorrect.
................
r97977 | russell | 2008-01-10 16:33:24 -0700 (Thu, 10 Jan 2008) | 11 lines
Merged revisions 97976 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines
Fix various timing calculations that made assumptions that the audio being
processed was at a sample rate of 8 kHz.
........
................
r97978 | tilghman | 2008-01-10 16:40:13 -0700 (Thu, 10 Jan 2008) | 14 lines
Merged revisions 97973 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines
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)
........
................
r98024 | russell | 2008-01-10 17:12:22 -0700 (Thu, 10 Jan 2008) | 3 lines
Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.
................
r98026 | russell | 2008-01-10 17:15:58 -0700 (Thu, 10 Jan 2008) | 10 lines
Blocked revisions 98025 via svnmerge
........
r98025 | russell | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines
Simplify this code with a suggestion from Luigi on the asterisk-dev list.
Instead of using is16kHz(), implement a format_rate() function.
........
................
r98027 | russell | 2008-01-10 17:38:23 -0700 (Thu, 10 Jan 2008) | 9 lines
Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you
to set the qualify frequency.
(closes issue #11597)
Reported by: wilder
Patches:
qualifyfreq5.patch uploaded by wilder (license 362)
-- with some mods by me
................
r98047 | mmichelson | 2008-01-10 17:54:54 -0700 (Thu, 10 Jan 2008) | 7 lines
Fix "core show translation" to not output information for "unknown" codecs.
This fix was made in favor of the proposed patch since it doesn't involve changing
a core codec define.
(closes issue #11722, reported and initially patched by caio1982, final patch by me)
................
r98081 | russell | 2008-01-10 20:37:19 -0700 (Thu, 10 Jan 2008) | 6 lines
- Fix the last set of places where incorrect assumptions were made about the
sample length with g722. It is _2_ samples per byte, not 1. This was all
over the place, and I believed it, and it is what caused me to take so long
to figure out what was broken.
- Update copyright information on codec_g722.
................
r98083 | russell | 2008-01-10 20:40:58 -0700 (Thu, 10 Jan 2008) | 9 lines
Blocked revisions 98082 via svnmerge
........
r98082 | russell | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines
Fix samples vs. length calculations for g722
........
................
r98124 | kpfleming | 2008-01-11 05:51:21 -0700 (Fri, 11 Jan 2008) | 3 lines
Ascom phones send Flash events as SIP INFO using '!' as the 'digit'
................
r98152 | tilghman | 2008-01-11 08:12:33 -0700 (Fri, 11 Jan 2008) | 2 lines
Documentation updates
................
r98193 | tilghman | 2008-01-11 09:08:43 -0700 (Fri, 11 Jan 2008) | 10 lines
Merged revisions 98164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines
Back out changes from revision 97077, since it wasn't perfect
........
................
r98218 | russell | 2008-01-11 10:17:54 -0700 (Fri, 11 Jan 2008) | 4 lines
At one point during working on this module, I had the lin/lin16 versions of the
framein callbacks different. However, they are now the same again, so remove
the duplicate code and use the same functions for the lin/lin16 versions.
................
r98220 | file | 2008-01-11 10:27:58 -0700 (Fri, 11 Jan 2008) | 12 lines
Merged revisions 98219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines
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
........
................
r98267 | russell | 2008-01-11 11:26:32 -0700 (Fri, 11 Jan 2008) | 18 lines
Blocked revisions 98265 via svnmerge
........
r98265 | russell | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines
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)
........
................
r98268 | tilghman | 2008-01-11 11:30:00 -0700 (Fri, 11 Jan 2008) | 2 lines
Commit Nick Gorham's suggestion for timestamp fix
................
r98269 | tilghman | 2008-01-11 11:35:30 -0700 (Fri, 11 Jan 2008) | 2 lines
Port Nick Gorham's timestamp patch to adaptive_odbc, too
................
r98270 | russell | 2008-01-11 11:48:07 -0700 (Fri, 11 Jan 2008) | 7 lines
Fix a bus error that happened when asterisk was built with optimizations on
with platforms that explode on unaligned access. I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on. If it makes
sense to you, feel free to enlighten me. :)
(closes issue #11725, patched by me)
................
r98308 | russell | 2008-01-11 12:05:24 -0700 (Fri, 11 Jan 2008) | 9 lines
Kevin noted that the thing that I _actually_ changed here was that I converted
a value from a double, to a float, back to a double. Sure enough, when I changed
my interim variable back to a double, it still blows up. Switching all of these
to a float fixes the problem. This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.
(related to issue #11725)
................
r98316 | mmichelson | 2008-01-11 12:12:05 -0700 (Fri, 11 Jan 2008) | 13 lines
Merged revisions 98315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines
Properly report the hangup cause as no answer when someone does not answer
(closes issue #10574, reported by boch, patched by moy)
........
................
r98318 | file | 2008-01-11 12:30:34 -0700 (Fri, 11 Jan 2008) | 14 lines
Merged revisions 98317 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines
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)
........
................
r98334 | file | 2008-01-11 12:53:01 -0700 (Fri, 11 Jan 2008) | 14 lines
Merged revisions 98325 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines
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)
........
................
r98400 | russell | 2008-01-11 15:52:41 -0700 (Fri, 11 Jan 2008) | 17 lines
Merged revisions 98390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines
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
........
................
r98424 | file | 2008-01-11 15:57:39 -0700 (Fri, 11 Jan 2008) | 2 lines
Testing something...
................
r98432 | file | 2008-01-11 15:59:13 -0700 (Fri, 11 Jan 2008) | 2 lines
I am no longer Rockin'
................
r98434 | file | 2008-01-11 16:00:21 -0700 (Fri, 11 Jan 2008) | 2 lines
drumkilla ftw.
................
r98435 | file | 2008-01-11 16:09:31 -0700 (Fri, 11 Jan 2008) | 2 lines
Goodbye again drumkilla.
................
r98436 | kpfleming | 2008-01-11 16:10:57 -0700 (Fri, 11 Jan 2008) | 11 lines
Add 'auto' signalling mode for Zaptel channels.
(closes issue #11690)
Reported by: tzafrir
Patches:
signaling_to_signalling.diff uploaded by tzafrir (license 46)
signalling_cleanup.diff uploaded by tzafrir (license 46)
zap_auto_default.diff uploaded by tzafrir (license 46)
zap_no_default_sig.diff uploaded by tzafrir (license 46)
zap_signal_auto.diff uploaded by tzafrir (license 46)
................
r98454 | russell | 2008-01-11 16:57:01 -0700 (Fri, 11 Jan 2008) | 9 lines
Add some extra checking to help out with a potential error when trying to
run "make asterisk.pdf" when not all of the right packages are installed.
(closes issue #10763)
Reported by: Corydon76
Patches:
20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
................
r98487 | tilghman | 2008-01-11 17:17:26 -0700 (Fri, 11 Jan 2008) | 12 lines
Merged revisions 98467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines
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)
........
................
r98488 | kpfleming | 2008-01-11 17:20:55 -0700 (Fri, 11 Jan 2008) | 8 lines
Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated.
(closes issue #11212)
Reported by: tzafrir
Patches:
zap_dnd.diff uploaded by tzafrir (modified by me) (license 46)
................
r98514 | pari | 2008-01-11 22:13:04 -0700 (Fri, 11 Jan 2008) | 1 line
merging a comment added in 1.4
................
r98536 | tilghman | 2008-01-12 11:12:56 -0700 (Sat, 12 Jan 2008) | 3 lines
Conversion to load manager.conf into memory did not convert the password
functions correctly. (Closes issue #11749)
................
[... 79681 lines stripped ...]
More information about the asterisk-commits
mailing list