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freeswitch/conf/freeswitch.xml
Anthony Minessale 9c79c2a3fb Add mod_event_socket remote client module and sample client. 
To Test:

uncomment or add from modules.conf
make installall again to compile it
uncomment the load line from freeswitch.xml

the default values are to bind to 127.0.0.1 port 8021

telnet to port 8021
enter "auth ClueCon" to authenticate

from here you can do the following:
*) events [xml|plain] <list of events to log or all for all>
*) noevents 
*) log <level> // same as the console.conf values
*) nolog
*) api <command> <arg>
*) exit

there is a perl client in scripts/socket called fs.pl

with the module up and loaded:
cd scripts/socket
perl fs.pl <optional log level>

you can enter a few api commands like "show or status"




git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2047 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-07-22 21:49:52 +00:00

446 lines
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<?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>
</configuration>
<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<!-- <load module="mod_syslog"/> -->
<!-- XML Interfaces -->
<!-- <load module="mod_xml_rpc"/> -->
<!-- Event Handlers -->
<!-- <load module="mod_event_multicast"/> -->
<!-- <load module="mod_event_test"/> -->
<!-- <load module="mod_zeroconf"/> -->
<!-- <load module="mod_xmpp_event"/> -->
<!-- <load module="mod_event_socket"/> -->
<!-- Directory Interfaces -->
<!-- <load module="mod_ldap"/> -->
<!-- Endpoints -->
<load module="mod_exosip"/>
<!--<load module="mod_iax"/>-->
<load module="mod_portaudio"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_wanpipe"/> -->
<!-- <load module="mod_dingaling"/> -->
<!-- Applications -->
<load module="mod_bridgecall"/>
<load module="mod_echo"/>
<load module="mod_dptools"/>
<!-- <load module="mod_ivrtest"/> -->
<load module="mod_playback"/>
<load module="mod_commands"/>
<!-- <load module="mod_commands"/> -->
<!-- Dialplan Interfaces -->
<load module="mod_dialplan_xml"/>
<!-- <load module="mod_dialplan_directory"/> -->
<!-- Codec Interfaces -->
<load module="mod_g711"/>
<load module="mod_gsm"/>
<load module="mod_l16"/>
<!-- <load module="mod_speex"/> -->
<!-- <load module="mod_ilbc"/> -->
<!-- File Format Interfaces -->
<load module="mod_sndfile"/>
<!-- Timers -->
<load module="mod_softtimer"/>
<!-- Languages -->
<!-- <load module="mod_spidermonkey"/> -->
<!-- <load module="mod_perl"/> -->
<!-- ASR /TTS -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Conference Bridges -->
<!--<load module="mod_conference"/>-->
</modules>
</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="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="exosip.conf" description="Exosip Endpoint">
<settings>
<param name="port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<!-- the @20 is optional number of ms you want to use. Use it only
if you know the codec supports it -->
<param name="codec-prefs" value="PCMU@20i,PCMA@20i"/>
<!-- Example to call for speex in wideband 16k mode
you can have up to 2 '@; after the codec name followed by either
'i' (interval eg 20i for 20ms) or 'k' (kilohertz eg 16000k for 16khz)-->
<!--<param name="codec-prefs" value="SPEEX@16000k"/>-->
<!-- Payload number to bind DTMF to-->
<param name="rfc2833-pt" value="101"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- auto sense NAT issues and adjust accordingly -->
<param name="use-rtp-auto-adjust" value="true"/>
<!-- pick one (default if not specified is 'guess'); -->
<param name="rtp-ip" value="guess"/>
<!-- <param name-"rtp-ip" value="10.0.0.1"/> -->
<!-- leave commented or 0.0.0.0 for all ip -->
<!-- <param name="sip-ip" value="127.0.0.1"/> -->
<!-- optional ; -->
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
<!-- specify 'myrealm' with certian key -->
<!-- use !myrealm! at beginning of url to activate -->
<!-- exosip/!myrealm!1000@dest -->
<!-- srtp:<param name="myrealm" value="ffffffffffffffffffffffffffffffffffffffffffffffffffffffffffff"/> -->
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
</settings>
</configuration>
<configuration name="woomera.conf" description="Woomera Endpoint">
<settings>
<param name="debug" value="0"/>
</settings>
</configuration>
<configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
<settings>
<param name="debug" value="1"/>
<param name="dialplan" value="XML"/>
<param name="mtu" value="320"/>
<param name="dtmf-on" value="800"/>
<param name="dtmf-off" value="100"/>
<param name="supress-dtmf-tone" value="yes"/>
</settings>
<span>
<param name="span" value="1"/>
<param name="node" value="cpe"/>
<!-- <param name="switch" value="ni2"/> -->
<param name="switch" value="dms100"/>
<!-- <param name="switch" value="lucent5e"/> -->
<!-- <param name="switch" value="att4ess"/> -->
<!-- <param name="switch" value="euroisdn"/> -->
<!-- <param name="switch" value="gr303eoc"/> -->
<!-- <param name="switch" value="gr303tmc"/> -->
<param name="dp" value="national"/>
<!-- <param name="dp" value="international"/> -->
<!-- <param name="dp" value="local"/> -->
<!-- <param name="dp" value="private"/> -->
<!-- <param name="dp" value="unknown"/> -->
<param name="l1" value="ulaw"/>
<!-- <param name="l1" value="alaw"/> -->
<param name="bchan" value="1-23"/>
<param name="dchan" value="24"/>
<param name="dialplan" value="XML"/>
</span>
</configuration>
<configuration name="portaudio.conf" description="Soundcard Endpoint">
<settings>
<param name="debug" value="2"/>
<param name="dialplan" value="XML"/>
<!-- partial string match on something in the name or the device # -->
<param name="indev" value="USB"/>
<param name="outdev" value="USB"/>
<param name="cid-name" value="FreeSwitch"/>
<param name="cid-num" value="5555551212"/>
</settings>
</configuration>
<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
<param name="publish" value="yes"/>
<param name="browse" value="_sip._udp"/>
</settings>
</configuration>
<configuration name="xmpp_event.conf" description="XMPP Event Handler">
<settings>
<param name="#debug" value="1"/>
<param name="jid" value="freeswitch@my.jabber.com/me"/>
<param name="passwd" value="mypass"/>
<param name="target-jid" value="freeswitch@reader.org/him"/>
</settings>
</configuration>
<configuration name="dialplan_directory.conf" description="Dialplan Directory">
<settings>
<param name="directory-name" value="ldap"/>
<param name="host" value="ldap.freeswitch.org"/>
<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
<param name="pass" value="test"/>
<param name="base" value="dc=freeswitch,dc=org"/>
</settings>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<!-- *NOTE* your resource (after the /) MUST contain the string "talk" (upper or lower case is ok) -->
<!-- *NOTE* as of May 2 2006 you must set"auto-login" to"true" if you want to be able to auto-login on startup"/> -->
<interface>
<param name="name" value="jingle"/>
<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"/>
<!-- 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"/>
</interface>
</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"/>
<!-- 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.server.com/gateway.cgi" bindings="configuration"/> -->
</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">
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
<profile name="default">
<!-- Sample Rate-->
<param name="rate" value="8000"/>
<!-- Number of milliseconds per frame -->
<param name="interval" value="20"/>
<!-- Energy level required for audio to be sent to the other users -->
<param name="energy-level" value="300"/>
<!-- TTS Engine to use -->
<!--<param name="tts-engine" value="cepstral"/>-->
<!-- TTS Voice to use -->
<!--<param name="tts-voice" value="david"/>-->
<!-- If TTS is enabled all audio-file params not beginning with '/'
will be considered text to say with TTS -->
<!-- File to play to acknowledge succees -->
<!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
<!-- File to play to acknowledge failure -->
<!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
<!-- File to play to acknowledge muted -->
<!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
<!-- File to play to acknowledge unmuted -->
<!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
<!-- File to play if you are alone in the conference -->
<!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
<!-- File to play when you join the conference -->
<!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
<!-- File to play when you leave the conference -->
<!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
<!-- File to play when you ae ejected from the conference -->
<!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
<!-- File to play when the conference is locked -->
<!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
<!-- File to play to prompt for a pin -->
<!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
<!-- File to play to when the pin is invalid -->
<!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="FreeSWITCH"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="8777423583"/>
</profile>
</profiles>
<rooms>
<room name="freeswitch" profile="default"/>
</rooms>
</configuration>
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<!-- Valid fields in conditions:
"dialplan, caller_id_name, ani, ani2, caller_id_number,
network_addr, rdnis, destination_number, uuid, source,
context, chan_name" -->
<!-- *NOTE* The special context name 'any' will match any context -->
<context name="default">
<extension name="tollfree">
<condition field="destination_number" expression="^(18[0{2}8{2}7{2}6{2}]\d{7})">
<action application="bridge" data="exosip/$1-freeswitch@voip.trxtel.com"/>
</condition>
</extension>
<extension name="devconf">
<condition field="destination_number" expression="^888$">
<action application="bridge" data="exosip/888@66.250.68.194"/>
</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:exosip/1234@66.250.68.194"/>
</condition>
</extension>
<!-- if the destination is an exact match on the extension name
you do not need any regex in the condition -->
<extension name="999">
<condition><action application="bridge" data="exosip/888@66.250.68.194"/></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_exosip"/>
<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_exosip"/>
<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/$1"/>
</condition>
</extension>
<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
<extension name="9999">
<condition field="source" expression="mod_iax"/>
<condition field="destination_number" expression="9999">
<action application="playback" data="/var/sounds/beep.gsm"/>
</condition>
</extension>
<!-- Call the FreeSWITCH conference via SIP -->
<extension name="FreeSWITCH Conference SIP">
<condition field="destination_number" expression="^888$">
<action application="bridge" data="exosip/888@66.250.68.194"/>
</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@66.250.68.194/888"/>
</condition>
</extension>
</context>
</section>
<section name="directory" description="User Directory">
</section>
</document>
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