freeswitch/conf/softphone/freeswitch.xml

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <X-PRE-PROCESS cmd="set" data="auto_answer=false"/>
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
  <X-PRE-PROCESS cmd="set" data="codec_prefs=CELT@48000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
  <X-PRE-PROCESS cmd="set" data="default_gateway=default"/>
  <X-PRE-PROCESS cmd="set" data="us-ring=%(2000, 4000, 440.0, 480.0)"/>
  <X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
  <X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>

  <section name="configuration" description="Various Configuration">
    <configuration name="cdr_csv.conf" description="CDR CSV Format">
      <settings>
	<param name="default-template" value="example"/>
	<param name="rotate-on-hup" value="true"/>
	<param name="legs" value="a"/>
      </settings>
      <templates>
	<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
      </templates>
    </configuration>

    <configuration name="console.conf" description="Console Logger">
      <mappings>
	<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
      </mappings>
      <settings>
	<param name="colorize" value="true"/>
	<param name="loglevel" value="$${console_loglevel}"/>
      </settings>
    </configuration>

    <configuration name="enum.conf" description="ENUM Module">
      <settings>
	<param name="default-root" value="e164.org"/>
	<param name="default-isn-root" value="freenum.org"/>
	<param name="query-timeout" value="10"/>
	<param name="auto-reload" value="true"/>
      </settings>
      <routes>
	<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/softphone/$1"/>
      </routes>
    </configuration>

    <configuration name="local_stream.conf" description="stream files from local dir">
      <directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
	<param name="rate" value="48000"/>
	<param name="shuffle" value="true"/>
	<param name="channels" value="1"/>
	<param name="interval" value="10"/>
	<param name="timer-name" value="soft"/>
      </directory>
    </configuration>

    <configuration name="logfile.conf" description="File Logging">
      <settings>
	<param name="rotate-on-hup" value="true"/>
      </settings>
      <profiles>
	<profile name="default">
	  <settings>
	  </settings>
	  <mappings>
	    <map name="all" value="debug,info,notice,warning,err,crit,alert"/>
	  </mappings>
	</profile>
      </profiles>
    </configuration>

    <configuration name="modules.conf" description="Modules">
      <modules>
	<load module="mod_console"/>
	<load module="mod_logfile"/>
	<load module="mod_enum"/>
	<load module="mod_cdr_csv"/>
	<load module="mod_portaudio"/>
	<load module="mod_sofia"/>
	<load module="mod_loopback"/>
	<load module="mod_commands"/>
	<load module="mod_dptools"/>
	<load module="mod_dialplan_xml"/>
	<load module="mod_voipcodecs"/>
	<load module="mod_celt"/>
	<load module="mod_sndfile"/>
	<load module="mod_tone_stream"/>
	<load module="mod_local_stream"/>
      </modules>
    </configuration>

    <configuration name="portaudio.conf" description="Soundcard Endpoint">
      <settings>
	<param name="indev" value=""/>
	<!-- device to use for output -->
	<param name="outdev" value=""/>
	<!--<param name="ringdev" value=""/>-->
	<param name="ring-file" value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
	<param name="ring-interval" value="5"/>
	<param name="hold-file" value="$${hold_music}"/>
	<!--<param name="timer-name" value="soft"/>-->
	<param name="dialplan" value="XML"/>
	<param name="cid-name" value="$${outbound_caller_name}"/>
	<param name="cid-num" value="$${outbound_caller_number}"/>
	<param name="sample-rate" value="48000"/>
	<param name="codec-ms" value="10"/>
      </settings>
    </configuration>

    <configuration name="post_load_modules.conf" description="Modules">
      <modules>
      </modules>
    </configuration>

    <configuration name="sofia.conf" description="sofia Endpoint">
      <global_settings>
	<param name="log-level" value="0"/>
	<param name="auto-restart" value="true"/>
	<param name="debug-presence" value="0"/>
      </global_settings>
      <profiles>
	<profile name="softphone">
	  <gateways>
	    <X-PRE-PROCESS cmd="include" data="accounts/*.xml"/>
	  </gateways>
	  <settings>
	    <!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
	    <param name="user-agent-string" value="FreeSWITCH/SoftPhone"/>
	    <!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
	    <param name="debug" value="0"/>
	    <param name="sip-trace" value="no"/>
	    <param name="context" value="public"/>
	    <param name="rfc2833-pt" value="101"/>
	    <!-- port to bind to for sip traffic -->
	    <param name="sip-port" value="12345"/>
	    <param name="dialplan" value="XML"/>
	    <param name="dtmf-duration" value="100"/>
	    <param name="codec-prefs" value="$${codec_prefs}"/>
	    <param name="use-rtp-timer" value="true"/>
	    <param name="rtp-timer-name" value="soft"/>
	    <!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
	    <param name="rtp-ip" value="$${local_ip_v4}"/>
	    <!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
	    <param name="sip-ip" value="$${local_ip_v4}"/>
	    <param name="hold-music" value="$${hold_music}"/>
	    <param name="apply-nat-acl" value="rfc1918"/>
	    <!--<param name="enable-timer" value="false"/>-->
	    <!--<param name="enable-100rel" value="true"/>-->
	    <!--<param name="minimum-session-expires" value="120"/>-->
	    <!--<param name="dtmf-type" value="info"/>-->
	    <param name="manage-presence" value="false"/>
	    <!--<param name="bitpacking" value="aal2"/> -->
	    <param name="max-proceeding" value="3"/>
	    <!--<param name="session-timeout" value="120"/>-->
	    <!--set to 'greedy' if you want your codec list to take precedence -->
	    <param name="inbound-codec-negotiation" value="generous"/>
	    <!-- if you want to send any special bind params of your own -->
	    <!--<param name="bind-params" value="transport=udp"/>-->
	    <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
	    <!--<param name="inbound-late-negotiation" value="true"/>-->
	    <!--<param name="accept-blind-reg" value="true"/> -->
	    <!--<param name="accept-blind-auth" value="true"/> -->
	    <!--<param name="suppress-cng" value="true"/> -->
	    <param name="nonce-ttl" value="60"/>
	    <!--<param name="NDLB-broken-auth-hash" value="true"/>-->
	    <!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
	    <param name="auth-calls" value="false"/>
	    <param name="auth-all-packets" value="false"/>
	    <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
	    <param name="ext-sip-ip" value="$${external_sip_ip}"/>
	    <!-- rtp inactivity timeout -->
	    <param name="rtp-timeout-sec" value="300"/>
	    <param name="rtp-hold-timeout-sec" value="1800"/>
	    <!-- VAD choose one (out is a good choice); -->
	    <!-- <param name="vad" value="in"/> -->
	    <!-- <param name="vad" value="out"/> -->
	    <!-- <param name="vad" value="both"/> -->
	    <param name="disable-register" value="true"/>
	    <!--<param name="NDLB-force-rport" value="true"/>-->
	    <param name="challenge-realm" value="auto_from"/>
	    <!--<param name="disable-rtp-auto-adjust" value="true"/>-->
	    <!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
	    <!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
	    <!--<param name="auto-rtp-bugs" data="clear"/>-->
	  </settings>
	</profile>
      </profiles>
    </configuration>

    <configuration name="switch.conf" description="Core Configuration">
      <cli-keybindings>
	<key name="1" value="help"/>
	<key name="2" value="status"/>
	<key name="3" value="pa answer"/>
	<key name="4" value="pa hangup"/>
	<key name="5" value="sofia status"/>
	<key name="6" value="reloadxml"/>
	<key name="7" value="console loglevel 0"/>
	<key name="8" value="console loglevel 7"/>
	<key name="9" value="sofia status profile softphone"/>
	<key name="10" value="fsctl pause"/>
	<key name="11" value="fsctl resume"/>
	<key name="12" value="version"/>
      </cli-keybindings> 
      <settings>
	<param name="colorize-console" value="true"/>
	<param name="max-sessions" value="20"/>
	<param name="sessions-per-second" value="5"/>
	<param name="loglevel" value="debug"/>
	<param name="crash-protection" value="false"/>
	<param name="dump-cores" value="yes"/>
	<param name="rtp-start-port" value="16384"/>
	<param name="rtp-end-port" value="16484"/>
      </settings>
    </configuration>
  </section>
  
  <section name="dialplan" description="Regex/XML Dialplan">
    <context name="default">
      <extension name="codec_and_sip_uri">
	<condition field="destination_number" expression="^sip:(.*):(.*)$">
	  <action application="bridge" data="{absolute_codec_string=$1}sofia/softphone/$2"/>
	</condition>
      </extension>
      <extension name="sip_uri">
	<condition field="destination_number" expression="^sip:(.*)$">
	  <action application="bridge" data="sofia/softphone/$1"/>
	</condition>
      </extension>
      <extension name="codec_and_number">
	<condition field="destination_number" expression="^(.*):(.*)@(.*)$">
	  <action application="bridge" data="{absolute_codec_string=$1}sofia/gateway/$3/$2"/>
	</condition>
      </extension>
      <extension name="number">
	<condition field="destination_number" expression="^(.*)@(.*)$">
	  <action application="bridge" data="sofia/gateway/$2/$1"/>
	</condition>
      </extension>
      <extension name="number">
	<condition field="destination_number" expression="^(.*)$">
	  <action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
	</condition>
      </extension>
    </context>
    <context name="public">
      <extension name="public_extensions">
	<condition field="$${auto_answer}" expression="^true$"/>
	<condition field="destination_number" expression="^(.*)$">
	  <action application="info"/>
	  <action application="bridge" data="portaudio/auto_answer"/>
	</condition>
      </extension>
      <extension name="public_extensions">
	<condition field="${sip_to_params}" expression="intercom=true"/>
	<condition field="${alert_info}" expression="Ring;Answer"/>
	<condition field="destination_number" expression="^(.*)$">
	  <action application="info"/>
	  <action application="bridge" data="portaudio/auto_answer"/>
	</condition>
      </extension>
      <extension name="public_extensions">
	<condition field="destination_number" expression="^(.*)$">
	  <action application="info"/>
	  <action application="set" data="ringback=${us-ring}"/>
	  <action application="pre_answer"/>
	  <action application="bridge" data="portaudio"/>
	</condition>
      </extension>
    </context>
  </section>
</document>