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authorJoshua Colp <jcolp@digium.com>2008-03-25 15:18:41 +0000
committerJoshua Colp <jcolp@digium.com>2008-03-25 15:18:41 +0000
commit738e4ec94e7144147b21c158764dc7d8fbbc20bf (patch)
treeb4442a43ce0035f8b86c6c864e171fcf650fca3c /CHANGES
parent358ac2f76a0313f1008627ab91e374e5bcaf4873 (diff)
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'CHANGES')
-rw-r--r--CHANGES5
1 files changed, 5 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index ba0ede212..bc224b4a8 100644
--- a/CHANGES
+++ b/CHANGES
@@ -26,6 +26,11 @@ Application Changes
of how many names are in your company. For large companies, this should be
quite helpful.
+SIP Changes
+-----------
+ * The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
+ audio file to be played upon completion of an attended transfer.
+
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--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
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