From ca6cf552f98949c5829f7ad8a2f0c84f2f32a791 Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Mon, 8 May 2006 15:46:02 +0000 Subject: Add documentation on "allowtransfer" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 3 +++ 1 file changed, 3 insertions(+) (limited to 'configs/sip.conf.sample') 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 -- cgit v1.2.3