<?xml version="1.0"?> <document type="freeswitch/xml"> <section name="configuration" description="Various Configuration"> <configuration name="switch.conf" description="Modules"> <settings> <!--Most channels to allow at once --> <param name="max-sessions" value="1000"/> </settings> <!--Any variables defined here will be available in every channel, in the dialplan etc --> <variables> <variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/> <variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/> <variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/> </variables> </configuration> <configuration name="modules.conf" description="Modules"> <modules> <!-- Loggers (I'd load these first) --> <load module="mod_console"/> <!-- <load module="mod_syslog"/> --> <!-- Multi-Faceted --> <!-- mod_enum is a dialplan interface, an application interface and an api command interface --> <load module="mod_enum"/> <!-- XML Interfaces --> <!-- <load module="mod_xml_rpc"/> --> <!-- <load module="mod_xml_curl"/> --> <!-- Event Handlers --> <!-- <load module="mod_cdr"/> --> <!-- <load module="mod_event_multicast"/> --> <!-- <load module="mod_event_socket"/> --> <!-- <load module="mod_xmpp_event"/> --> <!-- <load module="mod_zeroconf"/> --> <!-- Directory Interfaces --> <!-- <load module="mod_ldap"/> --> <!-- Endpoints --> <!-- <load module="mod_dingaling"/> --> <!--<load module="mod_iax"/>--> <load module="mod_portaudio"/> <load module="mod_sofia"/> <!-- <load module="mod_wanpipe"/> --> <!-- <load module="mod_woomera"/> --> <!-- Applications --> <load module="mod_bridgecall"/> <load module="mod_commands"/> <load module="mod_conference"/> <load module="mod_dptools"/> <load module="mod_echo"/> <!--<load module="mod_park"/>--> <load module="mod_playback"/> <!-- Dialplan Interfaces --> <!-- <load module="mod_dialplan_directory"/> --> <load module="mod_dialplan_xml"/> <!-- Codec Interfaces --> <load module="mod_g711"/> <load module="mod_gsm"/> <!-- <load module="mod_ilbc"/> --> <load module="mod_l16"/> <!-- <load module="mod_speex"/> --> <!-- File Format Interfaces --> <load module="mod_sndfile"/> <load module="mod_native_file"/> <!-- Timers --> <load module="mod_softtimer"/> <!-- Languages --> <!-- <load module="mod_spidermonkey"/> --> <!-- <load module="mod_perl"/> --> <!-- ASR /TTS --> <!-- <load module="mod_cepstral"/> --> <!-- <load module="mod_rss"/> --> </modules> </configuration> <configuration name="spidermonkey.conf" description="Spider Monkey JavaScript Plug-Ins"> <modules> <load module="mod_spidermonkey_teletone"/> <load module="mod_spidermonkey_core_db"/> <!--<load module="mod_spidermonkey_odbc"/>--> </modules> </configuration> <configuration name="event_multicast.conf" description="Multicast Event"> <settings> <param name="address" value="225.1.1.1"/> <param name="port" value="4242"/> <param name="bindings" value="all"/> </settings> </configuration> <configuration name="event_socket.conf" description="Socket Client"> <settings> <param name="listen-ip" value="127.0.0.1"/> <param name="listen-port" value="8021"/> <param name="password" value="ClueCon"/> </settings> </configuration> <configuration name="iax.conf" description="IAX Configuration"> <settings> <param name="debug" value="0"/> <!-- <param name="ip" value="1.2.3.4"> --> <param name="port" value="4569"/> <param name="dialplan" value="XML"/> <param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/> <param name="codec-master" value="us"/> <param name="codec-rates" value="8"/> </settings> </configuration> <configuration name="console.conf" description="Console Logger"> <!-- pick a file name, a function name or 'all' --> <!-- map as many as you need for specific debugging --> <mappings> <!-- <param name="log_event" value="DEBUG"/> --> <param name="all" value="DEBUG"/> </mappings> </configuration> <configuration name="sofia.conf" description="sofia Endpoint"> <profiles> <profile name="mydomain1.com"> <registrations> <!-- <registration name="asterlink"> <param name="register-scheme" value="Digest"/> <param name="register-realm" value=""/> <param name="register-username" value="1001"/> <param name="register-password" value="nhy65tgb"/> <param name="register-from" value="sip:1001@208.64.200.40"/> <param name="register-to" value="sip:1001@conference.freeswitch.org"/> <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/> <param name="register-frequency" value="20"/> </registration> --> </registrations> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU@20i"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="192.168.1.20"/> <param name="sip-ip" value="mydomain1.com"/> <!-- this lets anything register --> <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication --> <param name="accept-blind-reg" value="true"/> <!--<param name="auth-calls" value="true"/>--> <!-- on authed calls, authenticate *all* the packets not just invite --> <!--<param name="auth-all-packets" value="true"/>--> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- 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="alias" value="sip:10.0.1.251:5555"/>--> </settings> </profile> </profiles> </configuration> <configuration name="syslog.conf" description="Syslog Logger"> <!-- SYSLOG --> <!-- emerg - system is unusable --> <!-- alert - action must be taken immediately --> <!-- crit - critical conditions --> <!-- err - error conditions --> <!-- warning - warning conditions --> <!-- notice - normal, but significant, condition --> <!-- info - informational message --> <!-- debug - debug-level message --> <settings> <param name="ident" value="freeswitch"/> <param name="facility" value="user"/> <param name="format" value="${time} - ${message}"/> <param name="level" value="debug,info,warning-alert"/> </settings> </configuration> <configuration name="woomera.conf" description="Woomera Endpoint"> <settings> <param name="debug" value="0"/> </settings> <interface> <param name="host" value="localhost"/> <param name="port" value="42420"/> <param name="audio-ip" value="127.0.0.1"/> <param name="dialplan" value="XML"/> </interface> </configuration> <configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint"> <settings> <param name="debug" value="1"/> <param name="dialplan" value="XML"/> <param name="mtu" value="320"/> <param name="dtmf-on" value="800"/> <param name="dtmf-off" value="100"/> <param name="supress-dtmf-tone" value="yes"/> </settings> <span> <param name="span" value="1"/> <param name="node" value="cpe"/> <!-- <param name="switch" value="ni2"/> --> <param name="switch" value="dms100"/> <!-- <param name="switch" value="lucent5e"/> --> <!-- <param name="switch" value="att4ess"/> --> <!-- <param name="switch" value="euroisdn"/> --> <!-- <param name="switch" value="gr303eoc"/> --> <!-- <param name="switch" value="gr303tmc"/> --> <param name="dp" value="national"/> <!-- <param name="dp" value="international"/> --> <!-- <param name="dp" value="local"/> --> <!-- <param name="dp" value="private"/> --> <!-- <param name="dp" value="unknown"/> --> <param name="l1" value="ulaw"/> <!-- <param name="l1" value="alaw"/> --> <param name="bchan" value="1-23"/> <param name="dchan" value="24"/> <param name="dialplan" value="XML"/> </span> </configuration> <configuration name="portaudio.conf" description="Soundcard Endpoint"> <settings> <param name="debug" value="2"/> <param name="dialplan" value="XML"/> <!-- partial string match on something in the name or the device # --> <param name="indev" value="USB"/> <param name="outdev" value="USB"/> <param name="cid-name" value="FreeSwitch"/> <param name="cid-num" value="5555551212"/> </settings> </configuration> <configuration name="zeroconf.conf" description="Zeroconf Event Handler"> <settings> <param name="publish" value="yes"/> <param name="browse" value="_sip._udp"/> </settings> </configuration> <configuration name="xmpp_event.conf" description="XMPP Event Handler"> <settings> <param name="#debug" value="1"/> <param name="jid" value="freeswitch@my.jabber.com/me"/> <param name="passwd" value="mypass"/> <param name="target-jid" value="freeswitch@reader.org/him"/> </settings> </configuration> <configuration name="dialplan_directory.conf" description="Dialplan Directory"> <settings> <param name="directory-name" value="ldap"/> <param name="host" value="ldap.freeswitch.org"/> <param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/> <param name="pass" value="test"/> <param name="base" value="dc=freeswitch,dc=org"/> </settings> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <!-- *NOTE* change <x-profile></x-profile> to <profile></profile> to enable --> <!-- Client Profile (Original mode) --> <x-profile type="client"> <param name="name" value="mydomain.com"/> <param name="login" value="myjid@myserver.com/talk"/> <param name="password" value="mypass"/> <param name="dialplan" value="XML"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="10.0.0.1"/> <param name="auto-login" value="true"/> <param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <!--<param name="server" value="alternate.server.com"/>--> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- or --> <!-- <param name="rtp-ip" value="my_lan_ip"/> --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> --> <!-- default extension (if one cannot be determined) --> <param name="exten" value="888"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <param name="vad" value="both"/> </x-profile> <!-- Component (Server to Server Login) --> <x-profile type="component"> <!-- All traffic for *@sub.mydomain.com will come to you --> <param name="name" value="sub.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.server.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- "_auto_" means the extension will be automaticly set to the called jid --> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </x-profile> </configuration> <configuration name="xml_curl.conf" description="cURL XML Gateway"> <settings> <!-- The url to a gateway cgi that can generate xml similar to what's in this file only on-the-fly (leave it commented if you dont need it) --> <!-- one or more |-delim of configuration|directory|dialplan --> <!--<param name="gateway-url" value="http://www.mydomain.com/test.cgi" bindings="dialplan"/>--> <!-- set this to provide authentication credentials to the server --> <!--<param name="gateway-credentials" value="muser:mypass"/>--> </settings> </configuration> <configuration name="xml_rpc.conf" description="XML RPC"> <settings> <!-- The port where you want to run the http service (default 8080) --> <param name="http-port" value="8080"/> <!-- if all 3 of the following params exist all http traffic will require auth --> <param name="auth-realm" value="freeswitch"/> <param name="auth-user" value="freeswitch"/> <param name="auth-pass" value="works"/> </settings> </configuration> <configuration name="rss.conf" description="RSS Parser"> <feeds> <!-- Just download the files to wherever and refer to them here --> <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> --> <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> --> </feeds> </configuration> <!-- None of these paths are real if you want any of these options you need to really set them up --> <configuration name="conference.conf" description="Audio Conference"> <!-- Advertise certian presence on startup . --> <advertise> <room name="888@sub.mydomain.com" status="FreeSWITCH"/> </advertise> <!-- Profiles are collections of settings you can reference by name. --> <profiles> <profile name="default"> <!-- Domain (for presence) --> <param name="domain" value="sub.mydomain.com"/> <!-- Sample Rate--> <param name="rate" value="8000"/> <!-- Number of milliseconds per frame --> <param name="interval" value="20"/> <!-- Energy level required for audio to be sent to the other users --> <param name="energy-level" value="300"/> <!-- TTS Engine to use --> <!--<param name="tts-engine" value="cepstral"/>--> <!-- TTS Voice to use --> <!--<param name="tts-voice" value="david"/>--> <!-- If TTS is enabled all audio-file params not beginning with --> <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS --> <!-- File to play to acknowledge succees --> <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>--> <!-- File to play to acknowledge failure --> <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>--> <!-- File to play to acknowledge muted --> <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>--> <!-- File to play to acknowledge unmuted --> <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>--> <!-- File to play if you are alone in the conference --> <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>--> <!-- File to play when you join the conference --> <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>--> <!-- File to play when you leave the conference --> <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>--> <!-- File to play when you ae ejected from the conference --> <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>--> <!-- File to play when the conference is locked --> <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>--> <!-- File to play to prompt for a pin --> <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>--> <!-- File to play to when the pin is invalid --> <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>--> <!-- Conference pin --> <!--<param name="pin" value="12345"/>--> <!-- Default Caller ID Name for outbound calls --> <param name="caller-id-name" value="FreeSWITCH"/> <!-- Default Caller ID Number for outbound calls --> <param name="caller-id-number" value="8777423583"/> </profile> </profiles> </configuration> <configuration name="enum.conf" description="ENUM Module"> <settings> <param name="default-root" value="e164.org"/> </settings> <routes> <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/$1"/> </routes> </configuration> <configuration name="ivr.conf" description="IVR menus"> <menus> <menu name="main" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3"> <entry action="menu-exit" digits="*"/> <entry action="menu-sub" digits="2" param="menu2"/> <entry action="menu-exec-api" digits="3" param="api arg"/> <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> <entry action="menu-back" digits="5"/> <entry action="menu-call-transfer" digits="7" param="888"/> <entry action="menu-sub" digits="8" param="menu8"/>> </menu> <menu name="menu8" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3"> <entry action="menu-back" digits="#"/> <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> <entry action="menu-top" digits="*"/> </menu> <menu name="menu2" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3"> <entry action="menu-back" digits="#"/> <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> <entry action="menu-top" digits="*"/> </menu> </menus> </configuration> </section> <section name="dialplan" description="Regex/XML Dialplan"> <!-- Valid fields in conditions: --> <!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, --> <!-- rdnis, destination_number, uuid, source, context, chan_name" --> <!-- *NOTE* The special context name 'any' will match any context --> <context name="default"> <extension name="tollfree"> <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$"> <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> </document>