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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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version.h.
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--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.
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allowed, not nobody)
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compiles under OS X. And still works under linux.
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | 2 lines
Strip terminal sequences from the verbose messages
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(related to issue #11706)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11718)
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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we use.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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 ...
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- 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.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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).
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passed a null argument (channel).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan 2008) | 3 lines
use the \retval doxygen command properly
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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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)
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remove some unused code, add/clarify some comments.
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and also change the background from white to something
different to show that we can make use of fonts with transparent
background.
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The background of this particular image is transparent so we
can preserve the original background when we draw strings.
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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.
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r97206 | mmichelson | 2008-01-08 15:24:48 -0600 (Tue, 08 Jan 2008) | 3 lines
Some coding guidelines-related cleanup
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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)
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http://www.asterisk.org/doxygen/trunk/extref.html
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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)
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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)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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.
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