Add documentation on "allowtransfer"

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 84200a53ad5552f80d8d98b2fdd49b242a2869ad..d44b3c5007fe4bbabbf95c65a79fbdac587ee869 100644 (file)
--- 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