Age | Commit message (Collapse) | Author | Files | Lines |
|
(closes issue #14574)
Reported by: KNK
Patches:
audiohook_volume_fix.diff uploaded by KNK (license 545)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179291 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
also allow an audiohook to be detached if it has not been attached.
(closes issue #14414)
Reported by: bluecrow76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@173902 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
I don't know how this crept back in when I had already
fixed it earlier
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166162 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166092 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines
Add a tolerance period for sync-triggered audiohooks
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149205 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137082 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines
Bump up the debug level for a message.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130635 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines
Remove redundant logic
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130237 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines
Fix a typo in audiohook_read_frame_both.
While this change has not been proven to fix any
specific issue, it is incorrect and could cause
unforeseen problems.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130174 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines
If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute.
(closes issue #12296)
Reported by: jvandal
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@113297 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
developers to adjust the volume on a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110542 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines
Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait).
(closes issue #11945)
Reported by: xheliox
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108084 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103842 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103840 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103838 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
audiohooks. This causes an error when we attempt to destroy the lock later
when freeing the audiohook.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98581 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98432 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98424 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89511 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89425 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
this helps portability.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89109 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
free the frame we get in ast_write.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81858 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Closes issue #10654, patch by snuffy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81560 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
is actually there.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80157 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
listen and manipulate the audio going through a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78649 f38db490-d61c-443f-a65b-d21fe96a405b
|