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these two links. So, I'm adding them so they are available for reference later.
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r94831 | russell | 2007-12-27 09:16:56 -0600 (Thu, 27 Dec 2007) | 5 lines
Now that the contexts lock is a read/write lock, it should not be locked here
in ast_hint_state_changed(). This makes it get locked recursively which now
causes a deadlock.
(closes issue #11080, thanks to callguy for the access to a deadlocked machine)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines
Change ast_translator_best_choice() to only pay attention to audio formats.
This fixes a problem where Asterisk claims that a translation path can not be
found for channels involving video.
(closes issue #11638)
Reported by: cwhuang
Tested by: cwhuang
Patch suggested by cwhuang, with some additional changes by me.
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r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines
Use the constant that I really meant to use here ...
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- Doxygen fixes
Note:
- This application needs better documentation and a RESULT code in the
dialplan.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) | 2 lines
make this comment explain the situation in an even more explicit fashion
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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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fread is simpler and more portable, and there is no performance gain
in using mmap.
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used flags appear closer to the top.
Reported by: tzafrir
Patch by: tzafrir,tilghman
(Closes issue #11528)
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text a bit.
Closes issue #11564, patch by pnlarsson (with some extra cleanup by me).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) | 6 lines
Workaround for what is probably a glibc bug (but we'll see this crop up again
and again, if we don't add the workaround).
Reported by: rolek
Patch by: tilghman
(Closes issue #11601, closes issue #11426)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11096)
Patches:
pbx_config.c.patch uploaded by moy (license 222)
pbx_dundi.c.patch uploaded by moy (license 222)
pbx_gtkconsole.c.patch uploaded by moy (license 222)
pbx_loopback.c.patch uploaded by moy (license 222)
pbx_realtime.c.patch uploaded by moy (license 222)
pbx_spool.c.patch uploaded by moy (license 222)
app_adsiprog.c.patch uploaded by moy (license 222)
app_alarmreceiver.c.patch uploaded by moy (license 222)
app_amd.c.patch uploaded by moy (license 222)
app_authenticate.c.patch uploaded by moy (license 222)
app_cdr.c.patch uploaded by moy (license 222)
app_zapateller.c.patch uploaded by moy (license 222)
app_zapbarge.c.patch uploaded by moy (license 222)
app_zapras.c.patch uploaded by moy (license 222)
app_zapscan.c.patch uploaded by moy (license 222)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines
Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting
autoservice, remember it and ensure that the channel has the same setting when
autoservice gets stopped. (pointed out by d1mas, patched up by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the existence of a dialplan target.
(closes issue #11579)
Reported by: irroot
Patches:
func_dialplan2.c uploaded by irroot (license 52)
-- Additional changes by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines
When a channel is in autoservice, mark a flag on the channel that says that
we only care about the END of a digit. That way, no magic digit emulation stuff
will happen when all we're doing is queueing up END frames.
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code clarity
Reported by: dimas
Patch by: dimas
(Closes issue #11280)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) | 3 lines
Don't try to send a parked call back to itself.
(closes issue #11622, reported by djrodman, patched by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) | 5 lines
List include/asterisk/version.h as a .PHONY target because we want the commands
listed for this target to be executed regardless of whether the file exists or
not. This fixes having the version not up to date when running from svn.
(closes issue #11619, reported by plack, fixed by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines
Don't store DTMF BEGIN frames while a channel is in autoservice. It's just
going to make ast_read() do a lot of extra work when the channel comes back
out of autoservice.
(closes issue #11628, patched by me)
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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)
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(closes issue #11625, reported and patched by sergee)
Thank you very much to sergee for adding this new feature!
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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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(Closes issue #11161)
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character. Also, fix the documentation to match the code.
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Reported by: dror99
Patch by: tilghman
(Closes issue #11162)
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r94769 | file | 2007-12-24 22:27:08 -0400 (Mon, 24 Dec 2007) | 2 lines
file says... build on the builders.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines
Race: we need to wait to queue a NewChannel event until after the channel is
inserted into the channel list. The reason is because some manager users
immediately queue requests from the channel when they see that event and are
confused when Asterisk reports no such channel. (Closes issue #11632)
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r94765 | tilghman | 2007-12-24 10:17:01 -0600 (Mon, 24 Dec 2007) | 5 lines
More deadlock avoidance code (this time between sip_monitor and sip_hangup)
Reported by: apsaras
Patch by: tilghman
(Closes issue #11413)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007) | 5 lines
Another bit of bad logic in realtime_peer
Reported by: dimas
Patch by: dimas
(Closes issue #11631)
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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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app_voicemail.
Since the dtable in base_encode always gets populated with the same values every time and never
changes, make it static and const and only initialize it once. Also, there's no reason to
define BASEMAXINLINE twice, so remove the redundant #define.
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parameters, as well as RLT support for DMS type switches
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(closes issue #11610, reported and patched by rbraun_performatique)
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