Author: brian Date: Sat Dec 22 13:59:36 2007 New Revision: 6958 Removed: freeswitch/trunk/conf/dialplan/US.conf.xml Modified: freeswitch/trunk/conf/dialplan/default.xml freeswitch/trunk/conf/sip_profiles/default.xml Log: update default configs slightly Modified: freeswitch/trunk/conf/dialplan/default.xml ============================================================================== --- freeswitch/trunk/conf/dialplan/default.xml (original) +++ freeswitch/trunk/conf/dialplan/default.xml Sat Dec 22 13:59:36 2007 @@ -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}@$${domain}"/> Modified: freeswitch/trunk/conf/sip_profiles/default.xml ============================================================================== --- freeswitch/trunk/conf/sip_profiles/default.xml (original) +++ freeswitch/trunk/conf/sip_profiles/default.xml Sat Dec 22 13:59:36 2007 @@ -29,6 +29,8 @@ <param name="rtp-ip" value="$${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"/>