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| author | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-19 23:04:07 +0000 |
|---|---|---|
| committer | mmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-19 23:04:07 +0000 |
| commit | f616aed3d7d63dcfa37f669e63f51d3853d90d4a (patch) | |
| tree | 3b53ee1c38ccff9501d47d0f389bd4abfddf4246 /CHANGES | |
| parent | 811d9372de1ef4443c3372ec8cf9aeb6951470ec (diff) | |
Merged revisions 166092,166095 via svnmerge from
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/1.6.0@166097 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 4 |
1 files changed, 4 insertions, 0 deletions
@@ -92,6 +92,10 @@ Dialplan functions ID for the call (not the Asterisk call ID or unique ID), provided that the channel driver supports this. For SIP, you get the SIP call-ID for the bridged channel which you can store in the CDR with a custom field. + * Added the function AUDIOHOOK_INHERIT. This actually is already in Asterisk + 1.4, but since it was added late in the release cycle, I felt it was a good + idea to list it here as well. See the CLI output for "core show function + AUDIOHOOK_INHERIT" for more details CLI Changes ----------- |