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https://origsvn.digium.com/svn/asterisk/trunk
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r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines
Add Doxygen documentation for API changes from 1.6.0 to 1.6.1
Copied from my review board description:
This is a continuation of the API changes documentation started for describing
changes between releases. Most of the API changes were pretty simple needing
only to be brought to attention via the new "Asterisk API Changes" list.
However, if you see anything that needs further explanation feel free to
supplement what is there. The current method of documenting is to add (in the
header file): \version <ver number> <description of changes> and then to add
the function to the change list in doxyref.h on the AstAPIChanges page. I also
made sure all the functions that were newly added were tagged with \since
1.6.1. I think this is a good habit to start both for the historical aspect as
well as for the future ability to easily add a "New Asterisk API" page.
Review: http://reviewboard.digium.com/r/190/
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180740 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r166092 | mmichelson | 2008-12-19 16:26:16 -0600 (Fri, 19 Dec 2008) | 28 lines
Adding a new dialplan function AUDIOHOOK_INHERIT
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/
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r166095 | mmichelson | 2008-12-19 16:40:57 -0600 (Fri, 19 Dec 2008) | 5 lines
Remove the verbatim tag from the author line
I could have sworn I already did that before, though...
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166098 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines
Merged revisions 149204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@149206 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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developers to adjust the volume on a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110542 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108084 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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that a channel doesn't need to be locked before calling a certain function.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90141 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89519 f38db490-d61c-443f-a65b-d21fe96a405b
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Closes issue #10654, patch by snuffy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81560 f38db490-d61c-443f-a65b-d21fe96a405b
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functions. As inline functions, the lock debug information will show that
these are always locked in audiohooks.h instead of the file where the lock was
actually acquired.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81264 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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