update default configs slightly

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@6958 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/dialplan/US.conf.xml

-<context name="US-Numbering-Plan">
-
- <!--
- If the destination starts with 011, then check enum and route the call accordingly
- -->
- <extension name="US_International">
-   <condition field="destination_number" expression="^011(\d+)$">
-     <action application="set" data="continue_on_fail=true"/>
-     <action application="set" data="hangup_after_bridge=true"/>
-     <action application="enum" data="1$1"/>
-     <action application="bridge" data="${enum_auto_route}"/>
-     <action application="bridge" data="sofia/gateway/${default_gateway}/011$1"/>
-   </condition>
- </extension>
-
- <!--
- If the destination is a 10 or 11 digit US-Style number, then check enum and route accordingly
- -->
- <extension name="US_LD">
-   <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">
-     <action application="set" data="continue_on_fail=true"/>
-     <action application="set" data="hangup_after_bridge=true"/>
-     <action application="enum" data="1$1"/>
-     <action application="bridge" data="${enum_auto_route}"/>
-     <action application="bridge" data="sofia/gateway/${default_gateway}/1$1"/>
-   </condition>
- </extension>
-
- <!--
- Check for the existence of a default area code.... If found and the destination was dialed
- as 7 digits prefix a 1 and the default area code to the dialstring, then check enum,
- then (you guessed it) route accordingly
- -->
- <extension name="US_Local">
-   <condition field="${default_area_code}" expression="\d{3}" continue="on-true">
-     <anti-action application="speak" data="you must dial the area code to call this destination"/>
-   </condition>
-   <condition field="destination_number" expression="^([2-9]\d{6})$">
-     <action application="set" data="continue_on_fail=true"/>
-     <action application="set" data="hangup_after_bridge=true"/>
-     <action application="enum" data="1$1"/>
-     <action application="bridge" data="${enum_auto_route}"/>
-     <action application="bridge" data="sofia/gateway/${default_gateway}/1${default_area_code}$1"/>
-   </condition>
- </extension>
-
- <!--
- If the number dialed was
- 411 (Directory Services),
- 611 (Pretty Standard for Provider Customer Service),
- 811 (call before you dig i think?),
- or 911 (Emergency Services) then send directly out the default profile
- -->
- <extension name="FCC_Services">
-   <condition field="^([4689]11)$">
-     <action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
-   </condition>
- </extension>
-</context>

conf/dialplan/default.xml

@@ -113,8 +113,7 @@
 	<anti-action application="set" data="transfer_ringback=${us-ring}"/>
 	<anti-action application="set" data="call_timeout=30"/>
 	<anti-action application="set" data="hangup_after_bridge=true"/>
-	<anti-action application="set" data="left_hanging_extension=5900"/>
-	<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT"/>
+	<anti-action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,BUSY,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION"/>
 	<anti-action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
 	<anti-action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>
 	<anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/>

conf/sip_profiles/default.xml

@@ -29,6 +29,8 @@
     <param name="rtp-ip" value="$${local_ip_v4}"/>
     <param name="sip-ip" value="$${local_ip_v4}"/>
     <param name="hold-music" value="$${moh_uri}"/>
+    <!--<param name="dtmf-type" value="info"/>-->
+    <param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
     <!--enable to use presense and mwi -->
     <param name="manage-presence" value="true"/>
     <!-- This setting is for AAL2 bitpacking on G726 -->
@@ -44,13 +46,13 @@
     <!--<param name="bind-params" value="transport=udp"/>-->
 
     <!-- TLS: disabled by default, set to "true" to enable -->
-    <!--<param name="tls" value="false"/>-->
+    <param name="tls" value="true"/>
     <!-- additional bind parameters for TLS -->
-    <!--<param name="tls-bind-params" value="transport=tls"/>-->
+    <param name="tls-bind-params" value="transport=tls"/>
     <!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
-    <!--<param name="tls-sip-port" value="5061"/>-->
+    <param name="tls-sip-port" value="5061"/>
     <!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
-    <!--<param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>-->
+    <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
     
     <!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
     <!--<param name="rtp-rewrite-timestamps" value="true"/>-->
@@ -74,7 +76,7 @@
 	that the originator is using-->
     <!--<param name="disable-transcoding" value="true"/>-->
     <!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
-    <param name="NDLB-broken-auth-hash" value="true"/>
+    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
     <param name="auth-calls" value="true"/>
     <!-- on authed calls, authenticate *all* the packets not just invite -->
     <param name="auth-all-packets" value="false"/>