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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 /configs
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
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample5
1 files changed, 5 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index f64dc04a9..d54610686 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -476,6 +476,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; user options for whatever reason. The behavior is similar to
; how SIP URI's were typically handled in 1.6.2, hence the name.
+;send_diversion=no ; Default "yes" ; Asterisk normally sends Diversion headers with certain SIP
+ ; invites to relay data about forwarded calls. If this option
+ ; is disabled, Asterisk won't send Diversion headers unless
+ ; they are added manually.
+
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
; when this option is enabled. Disabling this option results in no modification