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This is disabled in the default build, you need to explicitly enable it
compiling with
make COPTS=-DHAVE_VIDEO_CONSOLE
In return, you will be able to do a video call with chan_oss, using
the webcam (or X11 grabbing) as local source, and rendering the
incoming stream on your screen. Currently supported formats are
h261, h263, h263+, h264, mpeg4 (all through the avcodec lib, part
of ffmpeg).
Incoming video is on the left, outgoing video is on the right,
while the center displays a keypad (if configured so).
Right clicking on the video windows increases the size,
center clicking reduces the size.
Dragging the mouse (with the left key) on the right window
while the X11 grabber is active moves the grab area.
This is the result of work by Sergio Fadda, Marta Carbone
and myself, all properly disclaimed to digium.
Note, there is a lot of work left to do in this module,
including adding support for Video4LinuxV2 (I have patches
from Matteo Brancaleoni which should be integrated),
and making the GUI a lot more friendly than it is now
(e.g. supporting merging or switching among multiple sources,
a text window, and more).
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Reported by: snuffy
Patch by: snuffy,tilghman
(Closes issue #11315)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 lines
Up the length of the format on the SIP channel since it can now be rather long.
(closes issue #11552)
Reported by: francesco_r
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 Dez 2007) | 1 line
fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread starting.
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Reported by: snuffy
Patch by: snuffy
(Closes issue #11547)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) | 5 lines
Properly initialize polarity statuses, so that they are detected properly.
Reported by: julianjm
Patch by: julianjm
(Closes issue #10238)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10690)
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r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines
If a typo is found in a config file, we previous continued on with what was already loaded.
We do not want to do this (see bug below for details).
This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.
Issue 10690.
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reported
issue #11449 has demonstrated that it actually was a performance hit on his
machine. I think that it is possible that it could still be a benefit on systems
under higher load, especially SMP systems, but I don't have enough time or interest
to find out at the moment.
(closes issue #11449)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11048)
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r92617 | qwell | 2007-12-12 15:15:45 -0600 (Wed, 12 Dec 2007) | 4 lines
Don't increment user count until after name has been recorded (if enabled).
Issue 11048, tested by pep.
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misdn stuff needs a lot of doxygenification
(Hello, Qwell :-) )
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structures missing. Patched configure to check for this stuff and
put a #ifdef around the offending code in chan_zap. Thanks to file
for overseeing this.
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the match..
Closes issue #11518.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 lines
Avoid reinvite race situations with two Asterisks trying
to reinvite each other in 1.4 and trunk.
This patch implements support for the 491 error code that
Asterisk 1.4 generates on situations where we get an
incoming INVITE and already has one in progress.
Thanks to mavetju for reporting and to Raj Jain for an
excellent explanation of the problem.
Patch by myself. Tested with 8 Asterisk servers connected
to each other in a training network.
Closes issue #10481
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generate loadable and embedded module lists.
Individual Makefiles now are a lot simpler, possibly as simple as this:
-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
MODULE_PREFIX=cdr_
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.
The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).
With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.
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the top level directory.
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Closes issue #11490
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when a D-channel goes down
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines
Add support for accepting and sending T.38 in the initial INVITE.
(closes issue #9402)
Reported by: thdei
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Action ZapShowChannels
Header changes
- Channel: -> ZapChannel
For active channels, the Channel: and Uniqueid: headers are added
You can now add a "ZapChannel: " argument to zapshowchannels actions
to only get information about one channel.
From the moremanager branch
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"Username" still works, but is deprecated.
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dependent on libpri.h
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so a reload would erase it and not reset it. (pointed out by tzafrir)
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print the debug message anyway if we can't find the right PRI
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my head.
(thanks to tzafrir for pointing it out)
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(Closes issue #11412)
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Closes issue #11464, patch by eliel.
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This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify.
The mwimonitor option enables MWI monitoring. When the MWI state on a line changes,
then the script specified by mwimonitornotify will be executed for custom handling
of the state change, similar to the externnotify option of voicemail.conf.
Also, when the MWI state on an FXO line changes, an internal Asterisk event is
generated to indicate the new state of the associated mailbox. That may, any
module that cares about MWI information will get notified and can handle it
just as if app_voicemail had sent this notification.
(BE-253, original patch from markster, with some minor modifications by me to
add comments, documentation, and internal event support)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec 2007) | 5 lines
Changing some bad logic when calculating the interdigit timeout.
(closes issue #11402, reported and patched by eferro)
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r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
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r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines
Fix locking issues under one legged replaces scenarios.
(closes issue #11420)
Reported by: irroot
Patches:
chan_sip_oneleg.patch uploaded by irroot (license 52)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov 2007) | 5 lines
Clear the DTMF buffer if the call times out.
(closes issue #11418, reported and patched by eferro)
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This patch is an optimization for chan_iax2. This module is now heavily
multi-threaded. However, there is still a good number of globally shared
resources that prevent things from happen asynchronously. One of those things
was the global IAX frame queue. This queue was used to hold frames that have
been deferred for transmitting by another thread, and frames that may need to
get retransmitted.
I changed the frame queue to be per-call, since almost all of the frame queue
handling only cares about frames specific to a call number.
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This replaces tab completion code with the use of a public function that
does the same thing
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follow.
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Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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Both still works in this version.
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