Author: anthm Date: Tue Jan 23 09:39:56 2007 New Revision: 4031 Added: freeswitch/trunk/conf/freeswitch_combined.xml Log: add original combined conf as an example Added: freeswitch/trunk/conf/freeswitch_combined.xml ============================================================================== --- (empty file) +++ freeswitch/trunk/conf/freeswitch_combined.xml Tue Jan 23 09:39:56 2007 @@ -0,0 +1,797 @@ +<?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"/> + + <!--Uncomment to set all inbound calls to no media mode--> + <!--<param name="inbound-no-media" 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="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> + +<!-- 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/$1"/> + </routes> + </configuration> + + <configuration name="ivr.conf" description="IVR menus"> + <menus> + <menu name="main" + greet-long="/soundfiles/greet-long.wav" + greet-short="/soundfiles/greet-short.wav" + invalid-sound="/soundfiles/invalid.wav" + exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3"> + <entry action="menu-exit" digits="*"/> + <entry action="menu-sub" digits="2" param="menu2"/> + <entry action="menu-exec-api" digits="3" param="api arg"/> + <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> + <entry action="menu-back" digits="5"/> + <entry action="menu-call-transfer" digits="7" param="888"/> + <entry action="menu-sub" digits="8" param="menu8"/>> + </menu> + <menu name="menu8" + greet-long="/soundfiles/greet-long.wav" + greet-short="/soundfiles/greet-short.wav" + invalid-sound="/soundfiles/invalid.wav" + exit-sound="/soundfiles/exit.wav" + timeout ="15" + max-failures="3"> + <entry action="menu-back" digits="#"/> + <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> + <entry action="menu-top" digits="*"/> + </menu> + <menu name="menu2" + greet-long="/soundfiles/greet-long.wav" + greet-short="/soundfiles/greet-short.wav" + invalid-sound="/soundfiles/invalid.wav" + exit-sound="/soundfiles/exit.wav" + timeout ="15" + max-failures="3"> + <entry action="menu-back" digits="#"/> + <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/> + <entry action="menu-top" digits="*"/> + </menu> + </menus> + </configuration> + + </section> + + <section name="dialplan" description="Regex/XML Dialplan"> + <!-- Valid fields in conditions: --> + <!-- "dialplan, caller_id_name, ani, ani2, caller_id_number, --> + <!-- rdnis, destination_number, uuid, source, context, chan_name" --> + + <!-- *NOTE* The special context name 'any' will match any context --> + <context name="default"> + <extension name="556"> <!-- demo phrases --> + <condition field="destination_number" expression="^556$"> + <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> + </section> + + <section name="directory" description="User Directory"> + <!--the domain or ip (the right hand side of the @ in the addr--> + <domain name="jabber.org"> + <!--the user id (the left hand side of the @ in the addr--> + <user id="stpeter"> + <params> + <!-- omit password for authless registration --> + <param name="password" value="mypass"/> + </params> + + <vcard xmlns='vcard-temp'> + <FN>Peter Saint-Andre</FN> + <N> + <FAMILY>Saint-Andre</FAMILY> + <GIVEN>Peter</GIVEN> + <MIDDLE/> + </N> + <NICKNAME>stpeter</NICKNAME> + <URL>http://www.jabber.org/people/stpeter.php</URL> + <BDAY>1966-08-06</BDAY> + <ORG> + <ORGNAME>Jabber Software Foundation</ORGNAME> + <ORGUNIT>Jabber Software Foundation</ORGUNIT> + </ORG> + <TITLE>Executive Director</TITLE> + <ROLE>Patron Saint</ROLE> + <TEL><WORK/><VOICE/><NUMBER>303-308-3282</NUMBER></TEL> + <TEL><WORK/><FAX/><NUMBER/></TEL> + <TEL><WORK/><MSG/><NUMBER/></TEL> + <ADR> + <WORK/> + <EXTADD>Suite 600</EXTADD> + <STREET>1899 Wynkoop Street</STREET> + <LOCALITY>Denver</LOCALITY> + <REGION>CO</REGION> + <PCODE>80202</PCODE> + <CTRY>USA</CTRY> + </ADR> + <TEL><HOME/><VOICE/><NUMBER>303-555-1212</NUMBER></TEL> + <TEL><HOME/><FAX/><NUMBER/></TEL> + <TEL><HOME/><MSG/><NUMBER/></TEL> + <ADR> + <HOME/> + <EXTADD/> + <STREET/> + <LOCALITY>Denver</LOCALITY> + <REGION>CO</REGION> + <PCODE>80209</PCODE> + <CTRY>USA</CTRY> + </ADR> + <EMAIL><INTERNET/><PREF/><USERID>stpeter@jabber.org</USERID></EMAIL> + <JABBERID>stpeter@jabber.org</JABBERID> + <DESC> + More information about me is located on my + personal website: http://www.saint-andre.com/ + </DESC> + </vcard> + + </user> + </domain> + </section> + + <!-- phrases section (under development still) --> + <section name="phrases" description="Speech Phrase Management"> + <macros> + <language name="en" sound_path="/snds" tts_engine="cepstral" tts_voice="david"> + <macro name="msgcount"> + <input pattern="(.*)"> + <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> + </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> + </language> + </macros> + </section> +</document> + +