freeswitch/fscomm/conf/freeswitch.xml
2010-01-14 04:54:03 +00:00

291 lines
12 KiB
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=$${local_ip_v4}"/>
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
<X-PRE-PROCESS cmd="set" data="codec_prefs=CELT@48000h,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
<X-PRE-PROCESS cmd="set" data="default_gateway=default"/>
<X-PRE-PROCESS cmd="set" data="us-ring=%(2000, 4000, 440.0, 480.0)"/>
<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
<section name="configuration" description="Various Configuration">
<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<param name="default-template" value="example"/>
<param name="rotate-on-hup" value="true"/>
<param name="legs" value="a"/>
</settings>
<templates>
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
</templates>
</configuration>
<configuration name="console.conf" description="Console Logger">
<mappings>
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
<settings>
<param name="colorize" value="true"/>
<param name="loglevel" value="$${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="qsettings.conf" description="configures our mod_qsettings">
<bindings value="configuration" />
</configuration>
<configuration name="local_stream.conf" description="stream files from local dir">
<directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
<param name="rate" value="48000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="10"/>
<param name="timer-name" value="soft"/>
</directory>
</configuration>
<configuration name="logfile.conf" description="File Logging">
<settings>
<param name="rotate-on-hup" value="true"/>
</settings>
<profiles>
<profile name="default">
<settings>
</settings>
<mappings>
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
</mappings>
</profile>
</profiles>
</configuration>
<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_logfile"/>
<load module="mod_enum"/>
<load module="mod_cdr_csv"/>
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<load module="mod_commands"/>
<load module="mod_dptools"/>
<load module="mod_dialplan_xml"/>
<load module="mod_voipcodecs"/>
<load module="mod_ilbc"/>
<load module="mod_speex"/>
<load module="mod_celt"/>
<load module="mod_siren"/>
<load module="mod_sndfile"/>
<load module="mod_tone_stream"/>
<load module="mod_local_stream"/>
<load module="mod_portaudio"/>
<load module="mod_event_socket"/>
</modules>
</configuration>
<configuration name="portaudio.conf" description="Soundcard Endpoint">
<settings>
<param name="indev" value=""/>
<!-- device to use for output -->
<param name="outdev" value=""/>
<!--<param name="ringdev" value=""/>-->
<param name="ring-file" value="tone_stream://%(2000,4000,440.0,480.0);loops=20"/>
<param name="ring-interval" value="5"/>
<param name="hold-file" value="$${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="event_socket.conf" description="Socket Client">
<settings>
<param name="nat-map" value="false"/>
<param name="listen-ip" value="127.0.0.1"/>
<param name="listen-port" value="8022"/>
<param name="password" value="ClueCon"/>
</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">
<settings>
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
<param name="user-agent-string" value="FreeSWITCH/FSComm"/>
<!-- <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>