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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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- 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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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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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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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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remove some unused code, add/clarify some comments.
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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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http://www.asterisk.org/doxygen/trunk/extref.html
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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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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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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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This way it can contain additional elements (e.g. fonts, buttons,
widgets) without having to use a zillion files to store them.
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minor code cleanups
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On passing, fix dialling from the keypad.
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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.
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svn repository to check out portaudio v19.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) | 4 lines
If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie.
Reported and patched by: one47
(Closes issue #11535)
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(closes issue #10393)
Reported by: tzafrir
Patches:
chan_alarm_asterisk.diff uploaded by tzafrir (license 46) (modified by me and added configure script support)
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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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r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | 3 lines
Don't crash if the iax2 pvt structure has been destroyed before we get to this point
(closes issue #11672, reported by snuffy, patched by me)
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indications can handle it
remove gentone and all the headers containing tones that are no longer needed
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remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
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r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line
when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense.
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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.
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... oops
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*should* work <G>
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properly
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if it is present, but doesn't parse any supplied parameters yet
(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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AST_STANDARD_APP_ARGS(foo, bar)
(closes issue #11668, reported and patched by mvanbaak)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines
Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001)
(closes issue #11637)
Reported by: greyvoip
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any platform as it was passing string pointers to a function expecting ints
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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.
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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).
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make it easier to add support for more grabbers (V4L2,
firewire, and so on).
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This makes it very convenient to enlarge images using the right-click
on the video window.
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need the buffer and mtu as input.
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