fun stuff

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9955 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/dialplan/default.xml

@@ -223,6 +223,7 @@
 	<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
 	<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
 	<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
+	<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>
 	<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
 	<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
 	<anti-action application="answer"/>

conf/directory/default/brian.xml

@@ -60,7 +60,8 @@
       <variable name="effective_caller_id_number" value="1000"/>
       <!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
       <variable name="process_cdr" value="true"/>
-
+      <!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
+      <variable name="sip_secure_media" value="true"/>
       <!-- limit the max number of outgoing calls for this user -->
       <!--<variable name="max_calls" value="2"/>-->