Author: anthm Date: Mon Jan 22 20:12:47 2007 New Revision: 4026 Added: freeswitch/trunk/conf/conference.conf.xml freeswitch/trunk/conf/console.conf.xml freeswitch/trunk/conf/default_context.xml freeswitch/trunk/conf/dialplan_directory.conf.xml freeswitch/trunk/conf/dingaling.conf.xml freeswitch/trunk/conf/directory.xml freeswitch/trunk/conf/enum.conf.xml freeswitch/trunk/conf/event_multicast.conf.xml freeswitch/trunk/conf/event_socket.conf.xml freeswitch/trunk/conf/iax.conf.xml freeswitch/trunk/conf/ivr.conf.xml freeswitch/trunk/conf/lang_en.xml freeswitch/trunk/conf/lang_fr.xml freeswitch/trunk/conf/modules.conf.xml freeswitch/trunk/conf/portaudio.conf.xml freeswitch/trunk/conf/rss.conf.xml freeswitch/trunk/conf/sofia.conf.xml freeswitch/trunk/conf/spidermonkey.conf.xml freeswitch/trunk/conf/switch.conf.xml freeswitch/trunk/conf/syslog.conf.xml freeswitch/trunk/conf/wanpipe.conf.xml freeswitch/trunk/conf/woomera.conf.xml freeswitch/trunk/conf/xml_curl.conf.xml freeswitch/trunk/conf/xml_rpc.conf.xml freeswitch/trunk/conf/xmpp_event.conf.xml freeswitch/trunk/conf/zeroconf.conf.xml Modified: freeswitch/trunk/conf/freeswitch.xml freeswitch/trunk/src/switch_xml.c Log: xml preprocessor (calling all documentors and default config composers!!) Added: freeswitch/trunk/conf/conference.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/conference.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,84 @@ +<!-- 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> + + <!-- These are the default keys that map when you do not specify a caller control group --> + <!-- Note: none and default are reserved names for group names --> + <caller-controls> + <group name="default"> + <control action="mute" digits="0"/> + <control action="deaf mute" digits="*"/> + <control action="energy up" digits="9"/> + <control action="energy equ" digits="8"/> + <control action="energy dn" digits="7"/> + <control action="vol talk up" digits="3"/> + <control action="vol talk zero" digits="2"/> + <control action="vol talk dn" digits="1"/> + <control action="vol listen up" digits="6"/> + <control action="vol listen zero" digits="5"/> + <control action="vol listen dn" digits="4"/> + <control action="hangup" digits="#"/> + </group> + </caller-controls> + + <!-- Profiles are collections of settings you can reference by name. --> + <profiles> + <!--If no profile is specified it will default to "default"--> + <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"/> + <!-- Name of the caller control group to use for this profile --> + <!-- <param name="caller-controls" value="some name"/> --> + <!-- 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 beginning with --> + <!-- 'say:' will be considered text to say with TTS --> + <!-- Set a default path here so you can use relative paths in the other sound params--> + <!--<param name="sound-prefix" value="/soundfiles"/>--> + <!-- File to play to acknowledge succees --> + <!--<param name="ack-sound" value="beep.wav"/>--> + <!-- File to play to acknowledge failure --> + <!--<param name="nack-sound" value="beeperr.wav"/>--> + <!-- File to play to acknowledge muted --> + <!--<param name="muted-sound" value="muted.wav"/>--> + <!-- File to play to acknowledge unmuted --> + <!--<param name="unmuted-sound" value="unmuted.wav"/>--> + <!-- File to play if you are alone in the conference --> + <!--<param name="alone-sound" value="yactopitc.wav"/>--> + <!-- File to play when you join the conference --> + <!--<param name="enter-sound" value="welcome.wav"/>--> + <!-- File to play when you leave the conference --> + <!--<param name="exit-sound" value="exit.wav"/>--> + <!-- File to play when you ae ejected from the conference --> + <!--<param name="kicked-sound" value="kicked.wav"/>--> + <!-- File to play when the conference is locked --> + <!--<param name="locked-sound" value="locked.wav"/>--> + <!-- File to play when the conference is locked during the call--> + <!--<param name="is-locked-sound" value="is-locked.wav"/>--> + <!-- File to play when the conference is unlocked during the call--> + <!--<param name="is-unlocked-sound" value="is-unlocked.wav"/>--> + <!-- File to play to prompt for a pin --> + <!--<param name="pin-sound" value="pin.wav"/>--> + <!-- File to play to when the pin is invalid --> + <!--<param name="bad-pin-sound" value="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> Added: freeswitch/trunk/conf/console.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/console.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,8 @@ +<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> Added: freeswitch/trunk/conf/default_context.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/default_context.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,105 @@ +<!-- 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,${caller_id_name}"/> + <action application="phrase" data="timespec,12:45:15"/> + <action application="phrase" data="saydate,0"/> + <action application="phrase" data="msgcount,130"/> + <action application="phrase" data="ip-addr,66.250.68.194"/> + <action application="phrase" data="saydate,$strepoch(2006-03-23 7:23)"/> + <!--<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> Added: freeswitch/trunk/conf/dialplan_directory.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/dialplan_directory.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,9 @@ +<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> Added: freeswitch/trunk/conf/dingaling.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/dingaling.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,53 @@ +<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="$${domain}"/> + <param name="login" value="myjid@myserver.com/talk"/> + <param name="password" value="mypass"/> + <param name="dialplan" value="XML"/> + <param name="message" value="Jingle all the way"/> + <param name="rtp-ip" value="auto"/> + <param name="auto-login" value="true"/> + <param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/> + <!-- SASL "plain" or "md5" --> + <param name="sasl" value="plain"/> + <!-- if the server where the jabber is hosted is not the same as the one in the jid --> + <!--<param name="server" value="alternate.server.com"/>--> + <!-- Enable TLS or not --> + <param name="tls" value="true"/> + <!-- disable to trade async for more calls --> + <param name="use-rtp-timer" value="true"/> + <!-- or --> + <!-- <param name="rtp-ip" value="auto"/> --> + <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> --> + <!-- default extension (if one cannot be determined) --> + <param name="exten" value="888"/> + <!-- VAD choose one --> + <!-- <param name="vad" value="in"/> --> + <!-- <param name="vad" value="out"/> --> + <param name="vad" value="both"/> + </x-profile> + + <!-- Component (Server to Server Login) --> + <x-profile type="component"> + <!-- All traffic for *@sub.mydomain.com will come to you --> + <param name="name" value="$${subdomain}"/> + <param name="password" value="secret"/> + <param name="dialplan" value="XML"/> + <param name="rtp-ip" value="auto"/> + <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> Added: freeswitch/trunk/conf/directory.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/directory.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,59 @@ +<!--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> Added: freeswitch/trunk/conf/enum.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/enum.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,11 @@ +<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> Added: freeswitch/trunk/conf/event_multicast.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/event_multicast.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,8 @@ +<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> + Added: freeswitch/trunk/conf/event_socket.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/event_socket.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,7 @@ +<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> Modified: freeswitch/trunk/conf/freeswitch.xml ============================================================================== --- freeswitch/trunk/conf/freeswitch.xml (original) +++ freeswitch/trunk/conf/freeswitch.xml Mon Jan 22 20:12:47 2007 @@ -1,800 +1,66 @@ <?xml version="1.0"?> <document type="freeswitch/xml"> - + <!--#comment + All comments starting with #command will be preprocessed and never sent to the xml parser + Valid instructions: + #include ==> Include another file to this exact point + (partial xml should be encased in <include></include> tags) + #set ==> Set a global variable (can be expanded during preprocessing with $$ variables) + (note the double $$ which denotes preprocessor variables) + #comment ==> A general comment such as this + + The preprocessor will compuile the full xml document to ${prefix}/log/freeswitch.registry + Don't modify it while freeswitch is running cos it is mem mapped in most cases =D + --> + + <!--#set "domain=mydomain.com"--> + <!--#set "subdomain=sub.mydomain.com"--> + <!--#set "default_codecs=PCUM@20i"--> + <!--my domain is $${domain}--> <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="auto"/> - <param name="sip-ip" value="auto"/> - - <!--Uncomment to set all inbound calls to no media mode--> - <!--<param name="inbound-no-media" value="true"/>--> - - <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> - <!--<param name="inbound-late-negotiation" value="true"/>--> - - <!-- 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> - <!-- indev, outdev, ringdev: - partial case sensitive string match on something in the name - or the device number prefixed with # eg "#1" (or blank for default) --> - - <!-- device to use for input --> - <param name="indev" value=""/> - <!-- device to use for output --> - <param name="outdev" value=""/> - - <!--device to use for inbound ring --> - <!--<param name="ringdev" value=""/>--> - <!--File to play as the ring sound --> - <!--<param name="ring-file" value="/sounds/ring.wav"/>--> - <!--Number of seconds to pause between rings --> - <!--<param name="ring-interval" value="5"/>--> - - <!--file to play when calls are on hold--> - <!--<param name="hold-file" value="/sounds/holdmusic.wav"/>--> - <!--Timer to use for hold music (i'd leave this one commented)--> - <!--<param name="timer-name" value="soft"/>--> - - <!--Default dialplan and caller-id info --> - <param name="dialplan" value="XML"/> - <param name="cid-name" value="FreeSwitch"/> - <param name="cid-num" value="5555551212"/> - - <!--audio sample rate and interval --> - <param name="sample-rate" value="8000"/> - <param name="codec-ms" value="20"/> - </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="auto"/> - <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="auto"/> --> - <!-- <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="auto"/> - <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> - -<!-- These are the default keys that map when you do not specify a caller control group --> -<!-- Note: none and default are reserved names for group names --> - <caller-controls> - <group name="default"> - <control action="mute" digits="0"/> - <control action="deaf mute" digits="*"/> - <control action="energy up" digits="9"/> - <control action="energy equ" digits="8"/> - <control action="energy dn" digits="7"/> - <control action="vol talk up" digits="3"/> - <control action="vol talk zero" digits="2"/> - <control action="vol talk dn" digits="1"/> - <control action="vol listen up" digits="6"/> - <control action="vol listen zero" digits="5"/> - <control action="vol listen dn" digits="4"/> - <control action="hangup" digits="#"/> - </group> - </caller-controls> - - <!-- Profiles are collections of settings you can reference by name. --> - <profiles> - <!--If no profile is specified it will default to "default"--> - <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"/> - <!-- Name of the caller control group to use for this profile --> - <!-- <param name="caller-controls" value="some name"/> --> - <!-- 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 beginning with --> - <!-- 'say:' will be considered text to say with TTS --> - <!-- Set a default path here so you can use relative paths in the other sound params--> - <!--<param name="sound-prefix" value="/soundfiles"/>--> - <!-- File to play to acknowledge succees --> - <!--<param name="ack-sound" value="beep.wav"/>--> - <!-- File to play to acknowledge failure --> - <!--<param name="nack-sound" value="beeperr.wav"/>--> - <!-- File to play to acknowledge muted --> - <!--<param name="muted-sound" value="muted.wav"/>--> - <!-- File to play to acknowledge unmuted --> - <!--<param name="unmuted-sound" value="unmuted.wav"/>--> - <!-- File to play if you are alone in the conference --> - <!--<param name="alone-sound" value="yactopitc.wav"/>--> - <!-- File to play when you join the conference --> - <!--<param name="enter-sound" value="welcome.wav"/>--> - <!-- File to play when you leave the conference --> - <!--<param name="exit-sound" value="exit.wav"/>--> - <!-- File to play when you ae ejected from the conference --> - <!--<param name="kicked-sound" value="kicked.wav"/>--> - <!-- File to play when the conference is locked --> - <!--<param name="locked-sound" value="locked.wav"/>--> - <!-- File to play when the conference is locked during the call--> - <!--<param name="is-locked-sound" value="is-locked.wav"/>--> - <!-- File to play when the conference is unlocked during the call--> - <!--<param name="is-unlocked-sound" value="is-unlocked.wav"/>--> - <!-- File to play to prompt for a pin --> - <!--<param name="pin-sound" value="pin.wav"/>--> - <!-- File to play to when the pin is invalid --> - <!--<param name="bad-pin-sound" value="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> - + <!--#include "switch.conf.xml"--> + <!--#include "modules.conf.xml"--> + <!--#include "spidermonkey.conf.xml"--> + <!--#include "event_multicast.conf.xml"--> + <!--#include "event_socket.conf.xml"--> + <!--#include "iax.conf.xml"--> + <!--#include "console.conf.xml"--> + <!--#include "sofia.conf.xml"--> + <!--#include "syslog.conf.xml"--> + <!--#include "woomera.conf.xml"--> + <!--#include "wanpipe.conf.xml"--> + <!--#include "portaudio.conf.xml"--> + <!--#include "zeroconf.conf.xml"--> + <!--#include "xmpp_event.conf.xml"--> + <!--#include "dialplan_directory.conf.xml"--> + <!--#include "dingaling.conf.xml"--> + <!--#include "xml_curl.conf.xml"--> + <!--#include "xml_rpc.conf.xml"--> + <!--#include "rss.conf.xml"--> + <!--#include "conference.conf.xml"--> + <!--#include "enum.conf.xml"--> + <!--#include "ivr.conf.xml"--> </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="ip-addr,66.250.68.194"/> - <action application="phrase" data="saydate,$strepoch(2006-03-23 7:23)"/> - <!--<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> + <!--#include "default_context.xml"--> </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> + <!--#include "directory.xml"--> </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="(.*)"> - <match> - <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"/>--> - </match> - </input> - </macro> - <macro name="saydate"> - <input pattern="(.*)"> - <match> - <action function="say" data="$1" method="pronounced" type="current_date_time"/> - </match> - </input> - </macro> - <macro name="timespec"> - <input pattern="(.*)"> - <match> - <action function="say" data="$1" method="pronounced" type="time_measurement"/> - </match> - </input> - </macro> - <macro name="ip-addr"> - <input pattern="(.*)"> - <match> - <action function="say" data="$1" method="iterated" type="ip_address"/> - <action function="say" data="$1" method="pronounced" type="ip_address"/> - </match> - </input> - </macro> - <macro name="spell"> - <input pattern="(.*)"> - <match> - <action function="say" data="$1" method="pronounced" type="name_spelled"/> - </match> - </input> - </macro> - <macro name="spell-phonetic"> - <input pattern="(.*)"> - <match> - <action function="say" data="$1" method="pronounced" type="name_phonetic"/> - </match> - </input> - </macro> - <macro name="tts-timeleft"> - <!-- The parser will visit each <input> tag and execute the actions in <match> or <nomatch> depending on the pattern param --> - <!-- If the function "break" is encountered all parsing will cease --> - <input pattern="(\d+):(\d+)"> - <match> - <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/> - <action function="break"/> - </match> - <nomatch> - <action function="speak-text" data="That input was invalid."/> - </nomatch> - </input> - <input pattern="(\d+) min (\d+) sec"> - <match> - <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/> - </match> - <nomatch> - <action function="speak-text" data="That input was invalid."/> - </nomatch> - </input> - </macro> + <!--#include "lang_en.xml"--> </language> <language name="fr" sound_path="/var/sounds/lang/fr/jean" tts_engine="cepstral" tts_voice="jean-pierre"> - <macro name="msgcount"> - <input pattern="(.*)"> - <match> - <action function="play-file" data="tuas.wav"/> - <action function="say" data="$1" method="pronounced" type="items"/> - <action function="play-file" data="messages.wav"/> - </match> - </input> - </macro> - <macro name="timeleft"> - <input pattern="(\d+):(\d+)"> - <match> - <action function="speak-text" data="il y a $1 minutes et de $2 secondes de restant"/> - </match> - </input> - </macro> + <!--#include "lang_fr.xml"--> </language> </macros> </section> -</document> +</document> Added: freeswitch/trunk/conf/iax.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/iax.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,11 @@ +<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> Added: freeswitch/trunk/conf/ivr.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/ivr.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,39 @@ +<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> Added: freeswitch/trunk/conf/lang_en.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/lang_en.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,71 @@ +<include><!--This line will be ignored it's here to validate the xml and is optional --> + <macro name="msgcount"> + <input pattern="(.*)"> + <match> + <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"/>--> + </match> + </input> + </macro> + <macro name="saydate"> + <input pattern="(.*)"> + <match> + <action function="say" data="$1" method="pronounced" type="current_date_time"/> + </match> + </input> + </macro> + <macro name="timespec"> + <input pattern="(.*)"> + <match> + <action function="say" data="$1" method="pronounced" type="time_measurement"/> + </match> + </input> + </macro> + <macro name="ip-addr"> + <input pattern="(.*)"> + <match> + <action function="say" data="$1" method="iterated" type="ip_address"/> + <action function="say" data="$1" method="pronounced" type="ip_address"/> + </match> + </input> + </macro> + <macro name="spell"> + <input pattern="(.*)"> + <match> + <action function="say" data="$1" method="pronounced" type="name_spelled"/> + </match> + </input> + </macro> + <macro name="spell-phonetic"> + <input pattern="(.*)"> + <match> + <action function="say" data="$1" method="pronounced" type="name_phonetic"/> + </match> + </input> + </macro> + <macro name="tts-timeleft"> + <!-- The parser will visit each <input> tag and execute the actions in <match> or <nomatch> depending on the pattern param --> + <!-- If the function "break" is encountered all parsing will cease --> + <input pattern="(\d+):(\d+)"> + <match> + <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/> + <action function="break"/> + </match> + <nomatch> + <action function="speak-text" data="That input was invalid."/> + </nomatch> + </input> + <input pattern="(\d+) min (\d+) sec"> + <match> + <action function="speak-text" data="You have $1 minutes, $2 seconds remaining $strftime(%Y-%m-%d)"/> + </match> + <nomatch> + <action function="speak-text" data="That input was invalid."/> + </nomatch> + </input> + </macro> +</include><!--This line will be ignored it's here to validate the xml and is optional --> Added: freeswitch/trunk/conf/lang_fr.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/lang_fr.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,18 @@ +<include><!--This line will be ignored it's here to validate the xml and is optional --> +<macro name="msgcount"> + <input pattern="(.*)"> + <match> + <action function="play-file" data="tuas.wav"/> + <action function="say" data="$1" method="pronounced" type="items"/> + <action function="play-file" data="messages.wav"/> + </match> + </input> +</macro> +<macro name="timeleft"> + <input pattern="(\d+):(\d+)"> + <match> + <action function="speak-text" data="il y a $1 minutes et de $2 secondes de restant"/> + </match> + </input> +</macro> +</include><!--This line will be ignored it's here to validate the xml and is optional --> Added: freeswitch/trunk/conf/modules.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/modules.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,68 @@ +<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> Added: freeswitch/trunk/conf/portaudio.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/portaudio.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,33 @@ +<configuration name="portaudio.conf" description="Soundcard Endpoint"> + <settings> + <!-- indev, outdev, ringdev: + partial case sensitive string match on something in the name + or the device number prefixed with # eg "#1" (or blank for default) --> + + <!-- device to use for input --> + <param name="indev" value=""/> + <!-- device to use for output --> + <param name="outdev" value=""/> + + <!--device to use for inbound ring --> + <!--<param name="ringdev" value=""/>--> + <!--File to play as the ring sound --> + <!--<param name="ring-file" value="/sounds/ring.wav"/>--> + <!--Number of seconds to pause between rings --> + <!--<param name="ring-interval" value="5"/>--> + + <!--file to play when calls are on hold--> + <!--<param name="hold-file" value="/sounds/holdmusic.wav"/>--> + <!--Timer to use for hold music (i'd leave this one commented)--> + <!--<param name="timer-name" value="soft"/>--> + + <!--Default dialplan and caller-id info --> + <param name="dialplan" value="XML"/> + <param name="cid-name" value="FreeSwitch"/> + <param name="cid-num" value="5555551212"/> + + <!--audio sample rate and interval --> + <param name="sample-rate" value="8000"/> + <param name="codec-ms" value="20"/> + </settings> +</configuration> Added: freeswitch/trunk/conf/rss.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/rss.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,7 @@ +<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> Added: freeswitch/trunk/conf/sofia.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/sofia.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,54 @@ +<configuration name="sofia.conf" description="sofia Endpoint"> + <profiles> + <profile name="$${domain}"> + <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="$${default_codecs}"/> + <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="auto"/> + <param name="sip-ip" value="auto"/> + + <!--Uncomment to set all inbound calls to no media mode--> + <!--<param name="inbound-no-media" value="true"/>--> + + <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> + <!--<param name="inbound-late-negotiation" value="true"/>--> + + <!-- 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> Added: freeswitch/trunk/conf/spidermonkey.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/spidermonkey.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,7 @@ +<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> Added: freeswitch/trunk/conf/switch.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/switch.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,13 @@ +<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> + Added: freeswitch/trunk/conf/syslog.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/syslog.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,17 @@ +<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> Added: freeswitch/trunk/conf/wanpipe.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/wanpipe.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,31 @@ +<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> Added: freeswitch/trunk/conf/woomera.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/woomera.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,11 @@ +<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> Added: freeswitch/trunk/conf/xml_curl.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/xml_curl.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,11 @@ +<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> Added: freeswitch/trunk/conf/xml_rpc.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/xml_rpc.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,10 @@ +<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> Added: freeswitch/trunk/conf/xmpp_event.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/xmpp_event.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,8 @@ +<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> Added: freeswitch/trunk/conf/zeroconf.conf.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/zeroconf.conf.xml Mon Jan 22 20:12:47 2007 @@ -0,0 +1,6 @@ +<configuration name="zeroconf.conf" description="Zeroconf Event Handler"> + <settings> + <param name="publish" value="yes"/> + <param name="browse" value="_sip._udp"/> + </settings> +</configuration> Modified: freeswitch/trunk/src/switch_xml.c ============================================================================== --- freeswitch/trunk/src/switch_xml.c (original) +++ freeswitch/trunk/src/switch_xml.c Mon Jan 22 20:12:47 2007 @@ -781,13 +781,204 @@ return &root->xml; } +static switch_size_t read_line(int fd, char *buf, switch_size_t len) { + char c, *p; + int cur; + switch_size_t total = 0; + + p = buf; + while (total + sizeof(c) < len && (cur = read(fd, &c, sizeof(c))) > 0) { + total += cur; + *p++ = c; + if (c == '\n') { + break; + } + } + + *p++ = '\0'; + return total; +} + +static char *expand_vars(char *buf, char *ebuf, switch_size_t elen, switch_size_t *newlen) +{ + char *var, *val; + char *rp = buf; + char *wp = ebuf; + char *ep = ebuf + elen - 1; + + if (!(var = strstr(rp, "$${"))) { + *newlen = strlen(buf); + return buf; + } + + while(*rp && wp < ep) { + + if (*rp == '$' && *(rp+1) == '$' && *(rp+2) == '{') { + char *e = strchr(rp, '}'); + + if (e) { + rp += 3; + var = rp; + *e++ = '\0'; + rp = e; + if ((val = switch_core_get_variable(var))) { + char *p; + for(p = val; p && *p && wp <= ep; p++) { + *wp++ = *p; + } + } + } + + } + + *wp++ = *rp++; + } + *wp++ = '\0'; + *newlen = strlen(ebuf); + + return ebuf; + +} + +static int preprocess(const char *file, int new_fd, int rlevel) +{ + int old_fd, close_fd = -1; + char *new_file = NULL; + switch_size_t cur = 0, ml = 0; + char *q, *cmd, buf[2048], ebuf[8192]; + + if ((old_fd = open(file, O_RDONLY, 0)) < 0) { + return old_fd; + } + + if (rlevel > 100) { + return -1; + } + + if (new_fd < 0) { + if (!(new_file = switch_mprintf("%s/freeswitch.registry", SWITCH_GLOBAL_dirs.log_dir))) { + goto done; + } + + if ((new_fd = open(new_file, O_WRONLY | O_CREAT | O_TRUNC, 0)) < 0) { + goto done; + } + close_fd = new_fd; + } + + while((cur = read_line(old_fd, buf, sizeof(buf))) > 0) { + char *arg, *e; + char *bp = expand_vars(buf, ebuf, sizeof(ebuf), &cur); + + /* we ignore <include> or </include> for the sake of validators */ + if (strstr(buf, "<include>") || strstr(buf, "</include>")) { + continue; + } + + if (ml) { + if ((e = strstr(buf, "-->"))) { + ml = 0; + bp = e + 3; + cur = strlen(bp); + } else { + continue; + } + } + + if ((cmd = strstr(bp, "<!--#"))) { + write(new_fd, bp, cmd - bp); + if ((e = strstr(cmd, "-->"))) { + *e = '\0'; + e += 3; + write(new_fd, e, strlen(e)); + } else { + ml++; + } + + cmd += 5; + if ((e = strchr(cmd, '\r')) || (e = strchr(cmd, '\n'))) { + *e = '\0'; + } + + if ((arg = strchr(cmd, ' '))) { + *arg++ = '\0'; + if ((q = strchr(arg, '"'))) { + char *qq = q+1; + + if ((qq = strchr(qq, '"'))) { + *qq = '\0'; + arg = q+1; + } + } + + if (!strcasecmp(cmd, "set")) { + char *name = arg; + char *val = strchr(name, '='); + + if (val) { + char *ve = val++; + while(*val && *val == ' ') { + val++; + } + *ve-- = '\0'; + while(*ve && *ve == ' ') { + *ve-- = '\0'; + } + } + + if (name && val) { + switch_core_set_variable(name, val); + } + + } else if (!strcasecmp(cmd, "include")) { + char *fme = NULL, *ifile = arg; + + if (!switch_is_file_path(ifile)) { + fme = switch_mprintf("%s%s%s", SWITCH_GLOBAL_dirs.conf_dir, SWITCH_PATH_SEPARATOR, arg); + ifile = fme; + } + if (preprocess(ifile, new_fd, rlevel + 1) < 0) { + fprintf(stderr, "Error including %s (%s)\n", ifile, strerror(errno)); + } + switch_safe_free(fme); + } /* else NO OP */ + } + + continue; + } + + write(new_fd, bp, cur); + } + + close(old_fd); + + if (close_fd > -1) { + close(close_fd); + new_fd = open(new_file, O_RDONLY, 0); + } + + done: + + switch_safe_free(new_file); + + if (new_fd < 0) { + return old_fd; + } + + return new_fd; +} + // a wrapper for switch_xml_parse_fd that accepts a file name SWITCH_DECLARE(switch_xml_t) switch_xml_parse_file(const char *file) { - int fd = open(file, O_RDONLY, 0); - switch_xml_t xml = switch_xml_parse_fd(fd); + int fd = -1; + switch_xml_t xml = NULL; + + if ((fd = preprocess(file, -1, 0)) > -1) { + xml = switch_xml_parse_fd(fd); + close(fd); + } - if (fd >= 0) close(fd); return xml; }