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authorOlle Johansson <oej@edvina.net>2006-05-08 15:46:02 +0000
committerOlle Johansson <oej@edvina.net>2006-05-08 15:46:02 +0000
commitca6cf552f98949c5829f7ad8a2f0c84f2f32a791 (patch)
tree4c240d6ca9a29476e23bf95a500a5bfb32821bd0 /configs/sip.conf.sample
parent55c6781c9e00b3bbbe89bdb43b574c9aec922a42 (diff)
Add documentation on "allowtransfer"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample3
1 files changed, 3 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 84200a53a..d44b3c500 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -29,6 +29,8 @@ context=default ; Default context for incoming calls
; this can also be set to 'osp'
; if asterisk was compiled with OSP support.)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
+;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
+ ; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk"
; Realms MUST be globally unique according to RFC 3261
@@ -334,6 +336,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; restrictcid restrictcid
; allowoverlap allowoverlap
; allowsubscribe allowsubscribe
+; allowtransfer allowtransfer
; subscribecontext subscribecontext
; videosupport videosupport
; maxcallbitrate maxcallbitrate