forked from Mirrors/freeswitch
d53878f39d
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3830 d0543943-73ff-0310-b7d9-9358b9ac24b2
728 lines
27 KiB
XML
728 lines
27 KiB
XML
<?xml version="1.0"?>
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<document type="freeswitch/xml">
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<section name="configuration" description="Various Configuration">
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<configuration name="switch.conf" description="Modules">
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<settings>
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<!--Most channels to allow at once -->
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<param name="max-sessions" value="1000"/>
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</settings>
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<!--Any variables defined here will be available in every channel, in the dialplan etc -->
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<variables>
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<variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/>
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<variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/>
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<variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/>
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</variables>
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</configuration>
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<configuration name="modules.conf" description="Modules">
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<modules>
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<!-- Loggers (I'd load these first) -->
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<load module="mod_console"/>
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<!-- <load module="mod_syslog"/> -->
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<!-- Multi-Faceted -->
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<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
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<load module="mod_enum"/>
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<!-- XML Interfaces -->
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<!-- <load module="mod_xml_rpc"/> -->
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<!-- <load module="mod_xml_curl"/> -->
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<!-- Event Handlers -->
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<!-- <load module="mod_cdr"/> -->
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<!-- <load module="mod_event_multicast"/> -->
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<!-- <load module="mod_event_socket"/> -->
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<!-- <load module="mod_xmpp_event"/> -->
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<!-- <load module="mod_zeroconf"/> -->
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<!-- Directory Interfaces -->
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<!-- <load module="mod_ldap"/> -->
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<!-- Endpoints -->
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<!-- <load module="mod_dingaling"/> -->
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<!--<load module="mod_iax"/>-->
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<load module="mod_portaudio"/>
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<load module="mod_sofia"/>
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<!-- <load module="mod_wanpipe"/> -->
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<!-- <load module="mod_woomera"/> -->
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<!-- Applications -->
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<load module="mod_bridgecall"/>
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<load module="mod_commands"/>
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<load module="mod_conference"/>
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<load module="mod_dptools"/>
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<load module="mod_echo"/>
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<!--<load module="mod_park"/>-->
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<load module="mod_playback"/>
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<!-- Dialplan Interfaces -->
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<!-- <load module="mod_dialplan_directory"/> -->
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<load module="mod_dialplan_xml"/>
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<!-- Codec Interfaces -->
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<load module="mod_g711"/>
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<load module="mod_gsm"/>
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<!-- <load module="mod_ilbc"/> -->
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<load module="mod_l16"/>
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<!-- <load module="mod_speex"/> -->
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<!-- File Format Interfaces -->
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<load module="mod_sndfile"/>
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<load module="mod_native_file"/>
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<!-- Timers -->
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<load module="mod_softtimer"/>
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<!-- Languages -->
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<!-- <load module="mod_spidermonkey"/> -->
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<!-- <load module="mod_perl"/> -->
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<!-- ASR /TTS -->
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<!-- <load module="mod_cepstral"/> -->
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<!-- <load module="mod_rss"/> -->
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</modules>
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</configuration>
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<configuration name="spidermonkey.conf" description="Spider Monkey JavaScript Plug-Ins">
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<modules>
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<load module="mod_spidermonkey_teletone"/>
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<load module="mod_spidermonkey_core_db"/>
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<!--<load module="mod_spidermonkey_odbc"/>-->
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</modules>
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</configuration>
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<configuration name="event_multicast.conf" description="Multicast Event">
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<settings>
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<param name="address" value="225.1.1.1"/>
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<param name="port" value="4242"/>
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<param name="bindings" value="all"/>
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</settings>
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</configuration>
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<configuration name="event_socket.conf" description="Socket Client">
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<settings>
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<param name="listen-ip" value="127.0.0.1"/>
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<param name="listen-port" value="8021"/>
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<param name="password" value="ClueCon"/>
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</settings>
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</configuration>
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<configuration name="iax.conf" description="IAX Configuration">
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<settings>
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<param name="debug" value="0"/>
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<!-- <param name="ip" value="1.2.3.4"> -->
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<param name="port" value="4569"/>
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<param name="dialplan" value="XML"/>
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<param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
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<param name="codec-master" value="us"/>
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<param name="codec-rates" value="8"/>
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</settings>
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</configuration>
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<configuration name="console.conf" description="Console Logger">
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<!-- pick a file name, a function name or 'all' -->
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<!-- map as many as you need for specific debugging -->
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<mappings>
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<!-- <param name="log_event" value="DEBUG"/> -->
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<param name="all" value="DEBUG"/>
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</mappings>
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</configuration>
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<configuration name="sofia.conf" description="sofia Endpoint">
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<profiles>
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<profile name="mydomain1.com">
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<registrations>
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<!-- <registration name="asterlink">
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<param name="register-scheme" value="Digest"/>
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<param name="register-realm" value=""/>
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<param name="register-username" value="1001"/>
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<param name="register-password" value="nhy65tgb"/>
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<param name="register-from" value="sip:1001@208.64.200.40"/>
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<param name="register-to" value="sip:1001@conference.freeswitch.org"/>
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<param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
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<param name="register-frequency" value="20"/>
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</registration> -->
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</registrations>
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<settings>
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<param name="debug" value="1"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="PCMU@20i"/>
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<param name="codec-ms" value="20"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="rtp-ip" value="192.168.1.20"/>
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<param name="sip-ip" value="mydomain1.com"/>
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<param name="accept-blind-reg" value="true"/>
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<!--<param name="auth-calls" value="true"/>-->
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<!-- on authed calls, authenticate *all* the packets not just invite -->
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<!--<param name="auth-all-packets" value="true"/>-->
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<!-- optional ; -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
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<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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</settings>
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</profile>
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</profiles>
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</configuration>
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<configuration name="syslog.conf" description="Syslog Logger">
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<!-- SYSLOG -->
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<!-- emerg - system is unusable -->
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<!-- alert - action must be taken immediately -->
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<!-- crit - critical conditions -->
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<!-- err - error conditions -->
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<!-- warning - warning conditions -->
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<!-- notice - normal, but significant, condition -->
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<!-- info - informational message -->
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<!-- debug - debug-level message -->
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<settings>
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<param name="ident" value="freeswitch"/>
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<param name="facility" value="user"/>
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<param name="format" value="${time} - ${message}"/>
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<param name="level" value="debug,info,warning-alert"/>
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</settings>
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</configuration>
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<configuration name="woomera.conf" description="Woomera Endpoint">
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<settings>
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<param name="debug" value="0"/>
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</settings>
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<interface>
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<param name="host" value="localhost"/>
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<param name="port" value="42420"/>
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<param name="audio-ip" value="127.0.0.1"/>
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<param name="dialplan" value="XML"/>
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</interface>
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</configuration>
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<configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
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<settings>
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<param name="debug" value="1"/>
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<param name="dialplan" value="XML"/>
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<param name="mtu" value="320"/>
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<param name="dtmf-on" value="800"/>
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<param name="dtmf-off" value="100"/>
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<param name="supress-dtmf-tone" value="yes"/>
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</settings>
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<span>
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<param name="span" value="1"/>
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<param name="node" value="cpe"/>
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<!-- <param name="switch" value="ni2"/> -->
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<param name="switch" value="dms100"/>
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<!-- <param name="switch" value="lucent5e"/> -->
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<!-- <param name="switch" value="att4ess"/> -->
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<!-- <param name="switch" value="euroisdn"/> -->
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<!-- <param name="switch" value="gr303eoc"/> -->
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<!-- <param name="switch" value="gr303tmc"/> -->
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<param name="dp" value="national"/>
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<!-- <param name="dp" value="international"/> -->
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<!-- <param name="dp" value="local"/> -->
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<!-- <param name="dp" value="private"/> -->
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<!-- <param name="dp" value="unknown"/> -->
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<param name="l1" value="ulaw"/>
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<!-- <param name="l1" value="alaw"/> -->
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<param name="bchan" value="1-23"/>
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<param name="dchan" value="24"/>
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<param name="dialplan" value="XML"/>
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</span>
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</configuration>
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<configuration name="portaudio.conf" description="Soundcard Endpoint">
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<settings>
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<param name="debug" value="2"/>
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<param name="dialplan" value="XML"/>
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<!-- partial string match on something in the name or the device # -->
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<param name="indev" value="USB"/>
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<param name="outdev" value="USB"/>
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<param name="cid-name" value="FreeSwitch"/>
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<param name="cid-num" value="5555551212"/>
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</settings>
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</configuration>
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<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
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<settings>
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<param name="publish" value="yes"/>
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<param name="browse" value="_sip._udp"/>
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</settings>
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</configuration>
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<configuration name="xmpp_event.conf" description="XMPP Event Handler">
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<settings>
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<param name="#debug" value="1"/>
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<param name="jid" value="freeswitch@my.jabber.com/me"/>
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<param name="passwd" value="mypass"/>
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<param name="target-jid" value="freeswitch@reader.org/him"/>
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</settings>
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</configuration>
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<configuration name="dialplan_directory.conf" description="Dialplan Directory">
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<settings>
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<param name="directory-name" value="ldap"/>
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<param name="host" value="ldap.freeswitch.org"/>
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<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
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<param name="pass" value="test"/>
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<param name="base" value="dc=freeswitch,dc=org"/>
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</settings>
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</configuration>
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<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
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<settings>
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<param name="debug" value="0"/>
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<param name="codec-prefs" value="PCMU"/>
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</settings>
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<!-- *NOTE* change <x-profile></x-profile> to <profile></profile> to enable -->
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<!-- Client Profile (Original mode) -->
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<x-profile type="client">
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<param name="name" value="mydomain.com"/>
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<param name="login" value="myjid@myserver.com/talk"/>
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<param name="password" value="mypass"/>
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<param name="dialplan" value="XML"/>
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<param name="message" value="Jingle all the way"/>
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<param name="rtp-ip" value="10.0.0.1"/>
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<param name="auto-login" value="true"/>
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<param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/>
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<!-- SASL "plain" or "md5" -->
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<param name="sasl" value="plain"/>
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<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
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<!--<param name="server" value="alternate.server.com"/>-->
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<!-- Enable TLS or not -->
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<param name="tls" value="true"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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<!-- or -->
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<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
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<!-- default extension (if one cannot be determined) -->
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<param name="exten" value="888"/>
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<!-- VAD choose one -->
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<!-- <param name="vad" value="in"/> -->
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<!-- <param name="vad" value="out"/> -->
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<param name="vad" value="both"/>
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</x-profile>
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<!-- Component (Server to Server Login) -->
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<x-profile type="component">
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<!-- All traffic for *@sub.mydomain.com will come to you -->
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<param name="name" value="sub.mydomain.com"/>
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<param name="password" value="secret"/>
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<param name="dialplan" value="XML"/>
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<param name="rtp-ip" value="208.64.200.42"/>
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<param name="server" value="jabber.server.org:5347"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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<!-- "_auto_" means the extension will be automaticly set to the called jid -->
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<param name="exten" value="_auto_"/>
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<!--<param name="vad" value="both"/>-->
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</x-profile>
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</configuration>
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<configuration name="xml_curl.conf" description="cURL XML Gateway">
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<settings>
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<!-- The url to a gateway cgi that can generate xml similar to
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what's in this file only on-the-fly (leave it commented if you dont
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need it) -->
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<!-- one or more |-delim of configuration|directory|dialplan -->
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<!--<param name="gateway-url" value="http://www.mydomain.com/test.cgi" bindings="dialplan"/>-->
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<!-- set this to provide authentication credentials to the server -->
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<!--<param name="gateway-credentials" value="muser:mypass"/>-->
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</settings>
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</configuration>
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<configuration name="xml_rpc.conf" description="XML RPC">
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<settings>
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<!-- The port where you want to run the http service (default 8080) -->
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<param name="http-port" value="8080"/>
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<!-- if all 3 of the following params exist all http traffic will require auth -->
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<param name="auth-realm" value="freeswitch"/>
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<param name="auth-user" value="freeswitch"/>
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<param name="auth-pass" value="works"/>
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</settings>
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</configuration>
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<configuration name="rss.conf" description="RSS Parser">
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<feeds>
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<!-- Just download the files to wherever and refer to them here -->
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<!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
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<!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
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</feeds>
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</configuration>
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<!-- None of these paths are real if you want any of these options you need to really set them up -->
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<configuration name="conference.conf" description="Audio Conference">
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<!-- Advertise certian presence on startup . -->
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<advertise>
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<room name="888@sub.mydomain.com" status="FreeSWITCH"/>
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</advertise>
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<!-- These are the default keys that map when you do not specify a caller control group-->
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<caller-controls>
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<group name="default">
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<control action="mute" digits="0"/>
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<control action="deaf mute" digits="*"/>
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<control action="energy up" digits="9"/>
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<control action="energy equ" digits="8"/>
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<control action="energy dn" digits="7"/>
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<control action="vol talk up" digits="3"/>
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<control action="vol talk zero" digits="2"/>
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<control action="vol talk dn" digits="1"/>
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<control action="vol listen up" digits="6"/>
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<control action="vol listen zero" digits="5"/>
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<control action="vol listen dn" digits="4"/>
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<control action="hangup" digits="#"/>
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</group>
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</caller-controls>
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<!-- Profiles are collections of settings you can reference by name. -->
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<profiles>
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<profile name="default">
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<!-- Domain (for presence) -->
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<param name="domain" value="sub.mydomain.com"/>
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<!-- Sample Rate-->
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<param name="rate" value="8000"/>
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<!-- Number of milliseconds per frame -->
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<param name="interval" value="20"/>
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<!-- Energy level required for audio to be sent to the other users -->
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<param name="energy-level" value="300"/>
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<!-- Name of the caller control group to use for this profile -->
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<!-- <param name="caller-controls" value="some name"/> -->
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<!-- TTS Engine to use -->
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<!--<param name="tts-engine" value="cepstral"/>-->
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<!-- TTS Voice to use -->
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<!--<param name="tts-voice" value="david"/>-->
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<!-- If TTS is enabled all audio-file params not beginning with -->
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<!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
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<!-- File to play to acknowledge succees -->
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<!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
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<!-- File to play to acknowledge failure -->
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<!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
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<!-- File to play to acknowledge muted -->
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<!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
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<!-- File to play to acknowledge unmuted -->
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<!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
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<!-- File to play if you are alone in the conference -->
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<!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
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<!-- File to play when you join the conference -->
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<!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
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<!-- File to play when you leave the conference -->
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<!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
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<!-- File to play when you ae ejected from the conference -->
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<!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
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<!-- File to play when the conference is locked -->
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<!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
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<!-- File to play to prompt for a pin -->
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<!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
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<!-- File to play to when the pin is invalid -->
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<!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
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<!-- Conference pin -->
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<!--<param name="pin" value="12345"/>-->
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<!-- Default Caller ID Name for outbound calls -->
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<param name="caller-id-name" value="FreeSWITCH"/>
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<!-- Default Caller ID Number for outbound calls -->
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<param name="caller-id-number" value="8777423583"/>
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</profile>
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</profiles>
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</configuration>
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<configuration name="enum.conf" description="ENUM Module">
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<settings>
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<param name="default-root" value="e164.org"/>
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</settings>
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<routes>
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<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
|
|
<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
|
|
<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
|
|
</routes>
|
|
</configuration>
|
|
|
|
<configuration name="ivr.conf" description="IVR menus">
|
|
<menus>
|
|
<menu name="main"
|
|
greet-long="/soundfiles/greet-long.wav"
|
|
greet-short="/soundfiles/greet-short.wav"
|
|
invalid-sound="/soundfiles/invalid.wav"
|
|
exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
|
|
<entry action="menu-exit" digits="*"/>
|
|
<entry action="menu-sub" digits="2" param="menu2"/>
|
|
<entry action="menu-exec-api" digits="3" param="api arg"/>
|
|
<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
|
|
<entry action="menu-back" digits="5"/>
|
|
<entry action="menu-call-transfer" digits="7" param="888"/>
|
|
<entry action="menu-sub" digits="8" param="menu8"/>>
|
|
</menu>
|
|
<menu name="menu8"
|
|
greet-long="/soundfiles/greet-long.wav"
|
|
greet-short="/soundfiles/greet-short.wav"
|
|
invalid-sound="/soundfiles/invalid.wav"
|
|
exit-sound="/soundfiles/exit.wav"
|
|
timeout ="15"
|
|
max-failures="3">
|
|
<entry action="menu-back" digits="#"/>
|
|
<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
|
|
<entry action="menu-top" digits="*"/>
|
|
</menu>
|
|
<menu name="menu2"
|
|
greet-long="/soundfiles/greet-long.wav"
|
|
greet-short="/soundfiles/greet-short.wav"
|
|
invalid-sound="/soundfiles/invalid.wav"
|
|
exit-sound="/soundfiles/exit.wav"
|
|
timeout ="15"
|
|
max-failures="3">
|
|
<entry action="menu-back" digits="#"/>
|
|
<entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
|
|
<entry action="menu-top" digits="*"/>
|
|
</menu>
|
|
</menus>
|
|
</configuration>
|
|
|
|
</section>
|
|
|
|
<section name="dialplan" description="Regex/XML Dialplan">
|
|
<!-- Valid fields in conditions: -->
|
|
<!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, -->
|
|
<!-- rdnis, destination_number, uuid, source, context, chan_name" -->
|
|
|
|
<!-- *NOTE* The special context name 'any' will match any context -->
|
|
<context name="default">
|
|
<extension name="556"> <!-- demo phrases -->
|
|
<condition field="destination_number" expression="^556$">
|
|
<action application="answer"/>
|
|
<action application="sleep" data="1000"/>
|
|
<action application="phrase" data="spell,${caller_id_name}"/>
|
|
<action application="phrase" data="spell-phonetic,${caller_id_name}"/>
|
|
<action application="phrase" data="timespec,12:45:15"/>
|
|
<action application="phrase" data="saydate,0"/>
|
|
<action application="phrase" data="msgcount,130"/>
|
|
<!--<action application="phrase" data="timeleft,3:30"/>-->
|
|
</condition>
|
|
</extension>
|
|
|
|
<extension name="tollfree">
|
|
<condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
|
|
<action application="enum" data="$1"/>
|
|
<action application="bridge" data="${enum_auto_route}"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- Call the FreeSWITCH conference via SIP -->
|
|
<!--<extension name="FreeSWITCH Conference SIP">-->
|
|
<!--<condition field="destination_number" expression="^888$">-->
|
|
<!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
|
|
<!--</condition>-->
|
|
<!--</extension> -->
|
|
|
|
<!-- Call the FreeSWITCH conference via IAX -->
|
|
<!--<extension name="FreeSWITCH Conference IAX">-->
|
|
<!--<condition field="destination_number" expression="^8888$">-->
|
|
<!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
|
|
<!--</condition>-->
|
|
<!--</extension>-->
|
|
|
|
<extension name="testmusic">
|
|
<condition field="destination_number" expression="^1234$">
|
|
<!-- Request a certain tone/file to be played while you wait for the call to be answered-->
|
|
<action application="set" data="ringback=${us-ring}"/>
|
|
<!--<action application="set" data="ringback=/home/ring.wav"/>-->
|
|
<action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- Enter an existing conference -->
|
|
<extension name="1000">
|
|
<condition field="destination_number" expression="^1000$">
|
|
<action application="conference" data="freeswitch"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- Start a dynamic conference and call someone at the same time -->
|
|
<extension name="2000">
|
|
<condition field="destination_number" expression="^2000$">
|
|
<action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- extensions starting with 4, all the numbers after 4 form a numeric filename -->
|
|
<!-- continue="true" means keep looking for more extensions to match -->
|
|
<!-- *NOTE* The entire dialplan is parsed ONCE when the call starts -->
|
|
<!-- so any call info acquired after the various actions cannot -->
|
|
<!-- be taken into consideration. -->
|
|
|
|
<!-- The first match will play a beep and the second one plays -->
|
|
<!-- the desired file. This is for demo purposes both actions -->
|
|
<!-- could have been under the same <extension> tag as well. -->
|
|
<extension name="playsound1" continue="true">
|
|
<condition field="source" expression="mod_sofia"/>
|
|
<condition field="destination_number" expression="^4(\d+)">
|
|
<action application="playback" data="/var/sounds/beep.gsm"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<extension name="playsound2">
|
|
<condition field="source" expression="mod_sofia"/>
|
|
<condition field="destination_number" expression="^4(\d+)">
|
|
<action application="playback" data="/root/$1.raw"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- send everything with a certian RDNIS to Wanpipe ISDN -->
|
|
<extension name="To PRI">
|
|
<condition field="rdnis" expression="8881231234"/>
|
|
<condition field="destination_number" expression="(.*)">
|
|
<action application="bridge" data="wanpipe/a/a/$1"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
|
|
<extension name="9999">
|
|
<condition field="source" expression="mod_iax"/>
|
|
<condition field="destination_number" expression="9999">
|
|
<action application="playback" data="/var/sounds/beep.gsm"/>
|
|
</condition>
|
|
</extension>
|
|
|
|
</context>
|
|
</section>
|
|
|
|
<section name="directory" description="User Directory">
|
|
<!--the domain or ip (the right hand side of the @ in the addr-->
|
|
<domain name="jabber.org">
|
|
<!--the user id (the left hand side of the @ in the addr-->
|
|
<user id="stpeter">
|
|
<params>
|
|
<!-- omit password for authless registration -->
|
|
<param name="password" value="mypass"/>
|
|
</params>
|
|
|
|
<vcard xmlns='vcard-temp'>
|
|
<FN>Peter Saint-Andre</FN>
|
|
<N>
|
|
<FAMILY>Saint-Andre</FAMILY>
|
|
<GIVEN>Peter</GIVEN>
|
|
<MIDDLE/>
|
|
</N>
|
|
<NICKNAME>stpeter</NICKNAME>
|
|
<URL>http://www.jabber.org/people/stpeter.php</URL>
|
|
<BDAY>1966-08-06</BDAY>
|
|
<ORG>
|
|
<ORGNAME>Jabber Software Foundation</ORGNAME>
|
|
<ORGUNIT>Jabber Software Foundation</ORGUNIT>
|
|
</ORG>
|
|
<TITLE>Executive Director</TITLE>
|
|
<ROLE>Patron Saint</ROLE>
|
|
<TEL><WORK/><VOICE/><NUMBER>303-308-3282</NUMBER></TEL>
|
|
<TEL><WORK/><FAX/><NUMBER/></TEL>
|
|
<TEL><WORK/><MSG/><NUMBER/></TEL>
|
|
<ADR>
|
|
<WORK/>
|
|
<EXTADD>Suite 600</EXTADD>
|
|
<STREET>1899 Wynkoop Street</STREET>
|
|
<LOCALITY>Denver</LOCALITY>
|
|
<REGION>CO</REGION>
|
|
<PCODE>80202</PCODE>
|
|
<CTRY>USA</CTRY>
|
|
</ADR>
|
|
<TEL><HOME/><VOICE/><NUMBER>303-555-1212</NUMBER></TEL>
|
|
<TEL><HOME/><FAX/><NUMBER/></TEL>
|
|
<TEL><HOME/><MSG/><NUMBER/></TEL>
|
|
<ADR>
|
|
<HOME/>
|
|
<EXTADD/>
|
|
<STREET/>
|
|
<LOCALITY>Denver</LOCALITY>
|
|
<REGION>CO</REGION>
|
|
<PCODE>80209</PCODE>
|
|
<CTRY>USA</CTRY>
|
|
</ADR>
|
|
<EMAIL><INTERNET/><PREF/><USERID>stpeter@jabber.org</USERID></EMAIL>
|
|
<JABBERID>stpeter@jabber.org</JABBERID>
|
|
<DESC>
|
|
More information about me is located on my
|
|
personal website: http://www.saint-andre.com/
|
|
</DESC>
|
|
</vcard>
|
|
|
|
</user>
|
|
</domain>
|
|
</section>
|
|
|
|
<!-- phrases section (under development still) -->
|
|
<section name="phrases" description="Speech Phrase Management">
|
|
<macros>
|
|
<language name="en" sound_path="/snds" tts_engine="cepstral" tts_voice="david">
|
|
<macro name="msgcount">
|
|
<input pattern="(.*)">
|
|
<action function="execute" data="sleep(1000)"/>
|
|
<action function="play-file" data="vm-youhave.wav"/>
|
|
<action function="say" data="$1" method="pronounced" type="items"/>
|
|
<action function="play-file" data="vm-messages.wav"/>
|
|
<!-- or -->
|
|
<!--<action function="speak-text" data="you have $1 messages"/>-->
|
|
</input>
|
|
</macro>
|
|
<macro name="saydate">
|
|
<input pattern="(.*)">
|
|
<action function="say" data="$1" method="pronounced" type="current_date_time"/>
|
|
</input>
|
|
</macro>
|
|
<macro name="timespec">
|
|
<input pattern="(.*)">
|
|
<action function="say" data="$1" method="pronounced" type="time_measurement"/>
|
|
</input>
|
|
</macro>
|
|
<macro name="spell">
|
|
<input pattern="(.*)">
|
|
<action function="say" data="$1" method="pronounced" type="name_spelled"/>
|
|
</input>
|
|
</macro>
|
|
<macro name="spell-phonetic">
|
|
<input pattern="(.*)">
|
|
<action function="say" data="$1" method="pronounced" type="name_phonetic"/>
|
|
</input>
|
|
</macro>
|
|
<macro name="tts-timeleft">
|
|
<input pattern="(\d+):(\d+)">
|
|
<action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/>
|
|
</input>
|
|
</macro>
|
|
</language>
|
|
<language name="fr" sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral" tts_voice="jean-pierre">
|
|
<macro name="msgcount">
|
|
<input pattern="(.*)">
|
|
<action function="play-file" data="tuas.wav"/>
|
|
<action function="say" data="$1" method="pronounced" type="items"/>
|
|
<action function="play-file" data="messages.wav"/>
|
|
</input>
|
|
</macro>
|
|
<macro name="timeleft">
|
|
<input pattern="(\d+):(\d+)">
|
|
<action function="speak-text" data="il y a $1 minutes et de $2 secondes de restant"/>
|
|
</input>
|
|
</macro>
|
|
</language>
|
|
</macros>
|
|
</section>
|
|
</document>
|
|
|
|
|