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authorJonathan Rose <jrose@digium.com>2012-02-27 16:24:17 +0000
committerJonathan Rose <jrose@digium.com>2012-02-27 16:24:17 +0000
commit299dd5d4fc2f6617ee924112c1edf2cc071a160b (patch)
tree5913b25cb36aa7ca552c8534f540982e243cf917 /CHANGES
parent9ed6de9fd254d4b51c493a8841fa1368ea4c0949 (diff)
Adds an option to sip.conf that prevents diversion headers from being added.
send_diversion=no will prevent Diversion headers from being added to SIP requests. This doesn't prevent Diversion from being added with dialplan such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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1 files changed, 2 insertions, 0 deletions
diff --git a/CHANGES b/CHANGES
index b0a9eddbd..b72862606 100644
--- a/CHANGES
+++ b/CHANGES
@@ -65,6 +65,8 @@ SIP Changes
which set the force_rport and comedia options automatically if Asterisk
detects that an incoming SIP request crossed a NAT after being sent by
the remote endpoint.
+ * Adds an option send_diversion which can be disabled to prevent
+ diversion headers from automatically being added to invites.
Chan_local changes
------------------