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提交 f6514066 authored 作者: Ken Rice's avatar Ken Rice 提交者： Travis Cross
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conf: add config styles: curl, insideout, sbc, softphone

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conf/curl/autoload_configs/acl.conf.xml 0 → 100644
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<configuration name="acl.conf" description="Network Lists">
<network-lists>
<list name="dl-candidates" default="allow">
<node type="deny" cidr="10.0.0.0/8"/>
<node type="deny" cidr="172.16.0.0/12"/>
<node type="deny" cidr="192.168.0.0/16"/>
</list>
<list name="rfc1918" default="deny">
<node type="allow" cidr="10.0.0.0/8"/>
<node type="allow" cidr="172.16.0.0/12"/>
<node type="allow" cidr="192.168.0.0/16"/>
</list>
<list name="lan" default="allow">
<node type="deny" cidr="192.168.42.0/24"/>
<node type="allow" cidr="192.168.42.42/32"/>
</list>
<list name="strict" default="deny">
<node type="allow" cidr="208.102.123.124/32"/>
</list>
<!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
-->
<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
</list>
</network-lists>
</configuration>
conf/curl/autoload_configs/alsa.conf.xml 0 → 100644
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<configuration name="alsa.conf" description="Soundcard Endpoint">
<settings>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="N800 Alsa"/>
<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>
conf/curl/autoload_configs/cdr_csv.conf.xml 0 → 100644
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<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<!-- 'cdr-csv' will always be appended to log-base -->
<!--<param name="log-base" value="/var/log"/>-->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
<param name="rotate-on-hup" value="true"/>
<!-- may be a b or ab -->
<param name="legs" value="a"/>
</settings>
<templates>
<template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
<template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
<template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
</templates>
</configuration>
conf/curl/autoload_configs/conference.conf.xml 0 → 100644
浏览文件 @ f6514066
<!-- http://wiki.freeswitch.org/wiki/Mod_conference -->
<!-- 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 certain presence on startup . -->
<advertise>
<room name="3001@$${domain}" 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="$${domain}"/>
<!-- Sample Rate-->
<param name="rate" value="8000"/>
<!-- Number of milliseconds per frame -->
<param name="interval" value="20"/>
<!-- Energy level required for audio to be sent to the other users -->
<param name="energy-level" value="300"/>
<!--Can be | delim of waste|mute|deaf waste will always transmit data to each channel
even during silence -->
<!--<param name="member-flags" value="waste"/>-->
<!-- Name of the caller control group to use for this profile -->
<!-- <param name="caller-controls" value="some name"/> -->
<!-- TTS Engine to use -->
<!--<param name="tts-engine" value="cepstral"/>-->
<!-- TTS Voice to use -->
<!--<param name="tts-voice" value="david"/>-->
<!-- If TTS is enabled all audio-file params beginning with -->
<!-- 'say:' will be considered text to say with TTS -->
<!-- Set a default path here so you can use relative paths in the other sound params-->
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<!-- 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="conference/conf-muted.wav"/>
<!-- File to play to acknowledge unmuted -->
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<!-- File to play if you are alone in the conference -->
<param name="alone-sound" value="conference/conf-alone.wav"/>
<!-- File to play endlessly (nobody will ever be able to talk) -->
<!--<param name="perpetual-sound" value="perpetual.wav"/>-->
<!-- File to play when you're alone (music on hold)-->
<param name="moh-sound" value="$${hold_music}"/>
<!-- File to play when you join the conference -->
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<!-- File to play when you leave the conference -->
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<!-- File to play when you ae ejected from the conference -->
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<!-- File to play when the conference is locked -->
<param name="locked-sound" value="conference/conf-locked.wav"/>
<!-- File to play when the conference is locked during the call-->
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<!-- File to play when the conference is unlocked during the call-->
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<!-- File to play to prompt for a pin -->
<param name="pin-sound" value="conference/conf-pin.wav"/>
<!-- File to play to when the pin is invalid -->
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="comfort-noise-level" value="1400"/>
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<!-- Suppress start and stop talking events -->
<!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
<!-- enable comfort noise generation -->
<param name="comfort-noise" value="true"/>
<!-- Uncomment auto-record to toggle recording every conference call. -->
<!-- Another valid value is shout://user:pass@server.com/live.mp3 -->
<!--
<param name="auto-record" value="$${base_dir}/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
</profile>
<profile name="wideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!--<param name="tts-engine" value="flite"/>-->
<!--<param name="tts-voice" value="kal16"/>-->
</profile>
<profile name="ultrawideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="32000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!--<param name="tts-engine" value="flite"/>-->
<!--<param name="tts-voice" value="kal16"/>-->
</profile>
<!--
<profile name="cdquality">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="interval" value="10"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
</profile>
-->
</profiles>
</configuration>
conf/curl/autoload_configs/console.conf.xml 0 → 100644
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<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>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
See examples below
The following map is the default, which is all debug levels enabled:
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
Example: the following turns on debugging for error and critical levels only
<map name="all" value="err,crit"/>
NOTE: using map name="all" will override any other settings! If you
want a more specific set of console messages then you will need
to specify which files and/or functions you want to have debug
messages. One option is to turn on just the more critical
messages with map name="all", then specify the other types of
console messages you want to see for various files and functions.
Example: turn on ERROR, CRIT, ALERT for all modules, then specify other
levels for various modules and functions
<map name="all" value="err,crit,alert"/>
<map name="switch_loadable_module_process" value="all"/>
<map name="mod_local_stream.c" value="warning,debug"/>
<map name="mod_sndfile.c" value="warning,info,debug"/>
-->
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
<!--
You can use or modify this sample set of mappings. It turns on higher
level messages for all modules and then specifies extra lower level
messages for OpenZAP, Sofia, and switch core messages.
<map name="all" value="warning,err,crit,alert"/>
<map name="zap_analog.c" value="all"/>
<map name="zap_io.c" value="all"/>
<map name="zap_isdn.c" value="all"/>
<map name="zap_zt.c" value="all"/>
<map name="mod_openzap" value="all"/>
<map name="sofia.c" value="notice"/>
<map name="switch_core_state_machine.c" value="all"/>
-->
</mappings>
<settings>
<!-- comment or set to false for no color logging -->
<param name="colorize" value="true"/>
<param name="loglevel" value="$${console_loglevel}"/>
</settings>
</configuration>
conf/curl/autoload_configs/dialplan_directory.conf.xml 0 → 100644
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<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>
conf/curl/autoload_configs/dingaling.conf.xml 0 → 100644
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<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>
</configuration>
conf/curl/autoload_configs/enum.conf.xml 0 → 100644
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<configuration name="enum.conf" description="ENUM Module">
<settings>
<param name="default-root" value="e164.org"/>
<param name="default-isn-root" value="freenum.org"/>
<param name="query-timeout" value="10"/>
<param name="auto-reload" value="true"/>
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1"/>
<!--<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>-->
<!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
</routes>
</configuration>
conf/curl/autoload_configs/event_multicast.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="event_multicast.conf" description="Multicast Event">
<settings>
<param name="address" value="225.1.1.1"/>
<param name="port" value="4242"/>
<param name="bindings" value="all"/>
</settings>
</configuration>
conf/curl/autoload_configs/event_socket.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="listen-ip" value="127.0.0.1"/>
<param name="listen-port" value="8021"/>
<param name="password" value="ClueCon"/>
<!--<param name="apply-inbound-acl" value="lan"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/fax.conf.xml 0 → 100644
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<configuration name="fax.conf" description="FAX application configuration">
<settings>
<param name="use-ecm" value="true"/>
<param name="verbose" value="false"/>
<param name="disable-v17" value="false"/>
<param name="ident" value="SpanDSP Fax Ident"/>
<param name="header" value="SpanDSP Fax Header"/>
<param name="spool-dir" value="/tmp"/>
<param name="file-prefix" value="faxrx"/>
</settings>
</configuration>
conf/curl/autoload_configs/fifo.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="fifo.conf" description="FIFO Configuration">
<fifos>
<fifo name="cool_fifo@$${domain}" importance="0">
<!--<member timeout="60" simo="1" lag="20">user/1005@$${domain}</member>-->
</fifo>
</fifos>
</configuration>
conf/curl/autoload_configs/iax.conf.xml 0 → 100644
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<configuration name="iax.conf" description="IAX Configuration">
<settings>
<param name="debug" value="0"/>
<param name="ip" value="$${local_ip_v4}"/>
<param name="port" value="4569"/>
<param name="context" value="public"/>
<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>
conf/curl/autoload_configs/ivr.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="ivr.conf" description="IVR menus">
<menus>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="demo_ivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
timeout ="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/> <!-- MOH -->
<entry action="menu-sub" digits="4" param="demo_ivr_submenu"/> <!-- demo sub menu -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
<!-- Demo IVR, Sub Menu -->
<menu name="demo_ivr_submenu"
greet-long="phrase:demo_ivr_sub_menu"
greet-short="phrase:demo_ivr_sub_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
timeout="15000"
max-failures="3"
max-timeouts="3">
<entry action="menu-top" digits="*"/>
</menu>
</menus>
</configuration>
<!-- TTS sample; non-functional but it demonstrates say: and TTS -->
<!--
<menu name="demo3"
greet-long="say:Press 1 to join the conference, Press 2 to join the other conference"
greet-short="say:Press 1 to join the conference, Press 2 to join the other conference"
invalid-sound="say:invalid extension"
exit-sound="say:exit sound"
timeout ="15000"
max-failures="3">
<entry action="menu-exit" digits="*"/>
<entry action="menu-playback" digits="1" param="say:You pressed 1"/>
<entry action="menu-call-transfer" digits="2" param=
"1000"/>
<entry action="menu-call-transfer" digits="3" param="1001"/>
</menu>
-->
conf/curl/autoload_configs/java.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="java.conf" description="Java Plug-Ins">
<!-- Path to the Java 1.6 virtual machine to use -->
<javavm path="/usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so"/>
<!-- Options to pass to Java -->
<options>
<!-- Your class path (make sure freeswitch.jar is on it) -->
<option value="-Djava.class.path=$${base_dir}/scripts/freeswitch.jar"/>
<!-- Enable remote debugging -->
<option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000"/>
</options>
</configuration>
conf/curl/autoload_configs/limit.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="limit.conf" description="LIMIT Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/local_stream.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="local_stream.conf" description="stream files from local dir">
<directory name="moh/8000" path="$${base_dir}/sounds/music/8000">
<param name="rate" value="8000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/16000" path="$${base_dir}/sounds/music/16000">
<param name="rate" value="16000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/32000" path="$${base_dir}/sounds/music/32000">
<param name="rate" value="32000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<!--
<directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
<param name="rate" value="48000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="10"/>
<param name="timer-name" value="soft"/>
</directory>
-->
</configuration>
conf/curl/autoload_configs/logfile.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="logfile.conf" description="File Logging">
<settings>
<!-- true to auto rotate on HUP, false to open/close -->
<param name="rotate-on-hup" value="true"/>
</settings>
<profiles>
<profile name="default">
<settings>
<!-- File to log to -->
<!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
<!-- At this length in bytes rotate the log file (0 for never) -->
<!--<param name="rollover" value="10485760"/>-->
</settings>
<mappings>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
Please see comments in console.conf.xml for more information
-->
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
</mappings>
</profile>
</profiles>
</configuration>
conf/curl/autoload_configs/lua.conf.xml 0 → 100644
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<configuration name="lua.conf" description="LUA Configuration">
<settings>
<!--
Specify local directories that will be searched for LUA modules
These entries will be pre-pended to the LUA_CPATH environment variable
-->
<!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
<!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->
<!--
Specify local directories that will be searched for LUA scripts
These entries will be pre-pended to the LUA_PATH environment variable
-->
<!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
<!-- <param name="script-directory" value="$${base_dir}/scripts/?.lua"/> -->
<!--<param name="xml-handler-script" value="/dp.lua"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a lua script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.lua"/>-->
<!--<param name="startup-script" value="startup_script_2.lua"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/modules.conf.xml 0 → 100644
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<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_xml_curl"/>
</modules>
</configuration>
conf/curl/autoload_configs/opal.conf.xml 0 → 100644
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<configuration name="opal.conf" description="Opal Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>
conf/curl/autoload_configs/openmrcp.conf.xml 0 → 100644
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<!-- http://wiki.freeswitch.org/wiki/Mod_openmrcp#Configuration -->
<configuration name="openmrcp.conf" description="mod_openmrcp configuration">
<settings>
<param name="asr_default_profile" value="openmrcp-v2"/>
<param name="tts_default_profile" value="openmrcp-v2"/>
</settings>
<profiles>
<X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/>
</profiles>
</configuration>
conf/curl/autoload_configs/openzap.conf.xml 0 → 100644
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<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<pri_spans>
<span name="PRI_1">
<!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="5ess"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
<span name="PRI_2">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="5ess"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
<!-- one entry here per openzap span -->
<analog_spans>
<span id="1">
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
<param name="tonegroup" value="us"/>
<param name="digit-timeout" value="2000"/>
<param name="max-digits" value="11"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="enable-callerid" value="true"/>
<!-- regex to stop dialing when it matches -->
<!--<param name="dial-regex" value="5555"/>-->
<!-- regex to stop dialing when it does not match -->
<!--<param name="fail-dial-regex" value="^5"/>-->
</span>
</analog_spans>
</configuration>
conf/curl/autoload_configs/perl.conf.xml 0 → 100644
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<configuration name="perl.conf" description="PERL Configuration">
<settings>
<!--<param name="xml-handler-script" value="/tmp/xml.pl"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a perl script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--param name="startup-script" value="startup_script_1.pl"/-->
<!--param name="startup-script" value="startup_script_2.pl"/-->
</settings>
</configuration>
conf/curl/autoload_configs/pocketsphinx.conf.xml 0 → 100644
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<configuration name="pocketsphinx.conf" description="PocketSphinx ASR Configuration">
<settings>
<param name="threshold" value="400"/>
<param name="silence-hits" value="25"/>
<param name="listen-hits" value="1"/>
<param name="auto-reload" value="true"/>
<!--<param name="narrowband-model" value="communicator"/>-->
<!--<param name="wideband-model" value="wsj1"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/portaudio.conf.xml 0 → 100644
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<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="$${outbound_caller_name}"/>
<param name="cid-num" value="$${outbound_caller_id}"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="8000"/>
<param name="codec-ms" value="20"/>
</settings>
</configuration>
conf/curl/autoload_configs/post_load_modules.conf.xml 0 → 100644
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<configuration name="post_load_modules.conf" description="Modules">
<modules>
</modules>
</configuration>
conf/curl/autoload_configs/python.conf.xml 0 → 100644
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<configuration name="python.conf" description="PYTHON Configuration">
<settings>
<!--<param name="xml-handler-script" value="dp"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a py script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.py"/>-->
<!--<param name="startup-script" value="startup_script_2.py"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/rss.conf.xml 0 → 100644
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<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>
conf/curl/autoload_configs/shout.conf.xml 0 → 100644
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<configuration name="shout.conf" description="mod shout config">
<settings>
<!-- Don't change these unless you are insane -->
<!--<param name="decoder" value="i586"/>-->
<!--<param name="volume" value=".1"/>-->
<!--<param name="outscale" value="8192"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/sofia.conf.xml 0 → 100644
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<configuration name="sofia.conf" description="sofia Endpoint">
<global_settings>
<param name="log-level" value="0"/>
<!-- <param name="auto-restart" value="false"/> -->
<param name="debug-presence" value="0"/>
</global_settings>
<!--
The rabbit hole goes deep. This includes all the
profiles in the sip_profiles directory that is up
one level from this directory.
-->
<profiles>
<X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
</profiles>
</configuration>
conf/curl/autoload_configs/spidermonkey.conf.xml 0 → 100644
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<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_socket"/>
<!--<load module="mod_spidermonkey_odbc"/>-->
</modules>
</configuration>
conf/curl/autoload_configs/switch.conf.xml 0 → 100644
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<configuration name="switch.conf" description="Core Configuration">
<cli-keybindings>
<key name="1" value="$${function_key_1}"/>
<key name="2" value="$${function_key_2}"/>
<key name="3" value="$${function_key_3}"/>
<key name="4" value="$${function_key_4}"/>
<key name="5" value="$${function_key_5}"/>
<key name="6" value="$${function_key_6}"/>
<key name="7" value="$${function_key_7}"/>
<key name="8" value="$${function_key_8}"/>
<key name="9" value="$${function_key_9}"/>
<key name="10" value="$${function_key_10}"/>
<key name="11" value="$${function_key_11}"/>
<key name="12" value="$${function_key_12}"/>
</cli-keybindings>
<settings>
<!--Colorize the Console -->
<param name="colorize-console" value="true"/>
<!--Most channels to allow at once -->
<param name="max-sessions" value="1000"/>
<!--Most channels to create per second -->
<param name="sessions-per-second" value="30"/>
<!-- Default Global Log Level - value is one of debug,info,notice,warning,err,crit,alert -->
<param name="loglevel" value="debug"/>
<!--Try to catch any crashes that can be recoverable (in the context of a call)-->
<param name="crash-protection" value="false"/>
<!--<param name="max_dtmf_duration" value="192000"/>-->
<!--<param name="default_dtmf_duration" value="8000"/>-->
<!--
If you want to send out voicemail notifications via Windows you'll need to change the mailer-app
variable to the setting below:
<param name="mailer-app" value="msmtp"/>
Donot change mailer-app-args.
You will also need to download a sendmail clone for Windows (msmtp). This version works without issue:
http://msmtp.sourceforge.net/index.html. Download and copy the .exe to %winddir%\system32.
You'll need to create a small config file for smtp credentials (host name, authentication, tls, etc.) in
%USERPROFILE%\Application Data\ called "msmtprc.txt". Below is a sample copy of this file:
###################################
# The SMTP server of the provider.
account provider
host smtp.myisp.com
from john@myisp.com
auth login
user johndoe
password mypassword
# Set a default account
account default : provider
###################################
-->
<param name="mailer-app" value="sendmail"/>
<param name="mailer-app-args" value="-t"/>
<param name="dump-cores" value="yes"/>
<!--RTP port range -->
<!--<param name="rtp-start-port" value="16384"/>-->
<!--<param name="rtp-end-port" value="32768"/>-->
</settings>
</configuration>
conf/curl/autoload_configs/syslog.conf.xml 0 → 100644
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<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="loglevel" value="warning"/>
</settings>
</configuration>
conf/curl/autoload_configs/timezones.conf.xml 0 → 100644
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差异被折叠。
conf/curl/autoload_configs/voicemail.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="voicemail.conf" description="Voicemail">
<settings>
</settings>
<profiles>
<profile name="default">
<param name="file-extension" value="wav"/>
<param name="terminator-key" value="#"/>
<param name="max-login-attempts" value="3"/>
<param name="digit-timeout" value="10000"/>
<param name="min-record-len" value="3"/>
<param name="max-record-len" value="300"/>
<param name="tone-spec" value="%(1000, 0, 640)"/>
<param name="callback-dialplan" value="XML"/>
<param name="callback-context" value="default"/>
<param name="play-new-messages-key" value="1"/>
<param name="play-saved-messages-key" value="2"/>
<param name="main-menu-key" value="0"/>
<param name="config-menu-key" value="5"/>
<param name="record-greeting-key" value="1"/>
<param name="choose-greeting-key" value="2"/>
<param name="change-pass-key" value="6"/>
<param name="record-name-key" value="3"/>
<param name="record-file-key" value="3"/>
<param name="listen-file-key" value="1"/>
<param name="save-file-key" value="2"/>
<param name="delete-file-key" value="7"/>
<param name="undelete-file-key" value="8"/>
<param name="email-key" value="4"/>
<param name="pause-key" value="0"/>
<param name="restart-key" value="1"/>
<param name="ff-key" value="6"/>
<param name="rew-key" value="4"/>
<param name="record-silence-threshold" value="200"/>
<param name="record-silence-hits" value="2"/>
<param name="web-template-file" value="web-vm.tpl"/>
<!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
<!--<param name="record-sample-rate" value="11025"/>-->
<!-- the next two both must be set for this to be enabled
the extension is in the format of <dest> [<dialplan>] [<context>]
-->
<param name="operator-extension" value="operator XML default"/>
<param name="operator-key" value="9"/>
<param name="vmain-extension" value="vmain XML default"/>
<param name="vmain-key" value="*"/>
<!-- playback created files as soon as they were recorded by default -->
<!--<param name="auto-playback-recordings" value="true"/>-->
<email>
<param name="template-file" value="voicemail.tpl"/>
<param name="notify-template-file" value="notify-voicemail.tpl"/>
<!-- this is the format voicemail_time will have -->
<param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
<param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
</email>
<!--<param name="storage-dir" value="/tmp"/>-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="record-comment" value="Your Comment"/>-->
<!--<param name="record-title" value="Your Title"/>-->
<!--<param name="record-copyright" value="Your Copyright"/>-->
</profile>
</profiles>
</configuration>
conf/curl/autoload_configs/xml_cdr.conf.xml 0 → 100644
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<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
<settings>
<!-- the url to post to if blank web posting is disabled -->
<!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
<!-- optional: credentials to send to web server -->
<!-- <param name="cred" value="user:pass"/> -->
<!-- the total number of retries (not counting the first 'try') to post to webserver incase of failure -->
<!-- <param name="retries" value="2"/> -->
<!-- delay between retries in seconds, default is 5 seconds -->
<!-- <param name="delay" value="1"/> -->
<!-- optional: if not present we do not log every record to disk -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank value will default to ${prefix}/logs/xml_cdr -->
<param name="log-dir" value=""/>
<!-- optional: if not present we do log the b leg -->
<!-- true or false if we should create a cdr for the b leg of a call-->
<param name="log-b-leg" value="false"/>
<!-- optional: if not present, all filenames are the uuid of the call -->
<!-- true or false if a leg files are prefixed "a_" -->
<param name="prefix-a-leg" value="true"/>
<!-- encode the post data may be 'true' for url encoding, 'false' for no encoding or 'base64' for base64 encoding -->
<param name="encode" value="true"/>
<!-- optional: set to true to disable Expect: 100-continue lighttpd requires this setting -->
<!--<param name="disable-100-continue" value="true"/>-->
<!-- optional: full path to the error log dir for failed web posts if not specified its the same as log-dir -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
<!-- <param name="err-log-dir" value="/tmp"/> -->
<!-- optional: if enabled this will disable CA root certificate checks by libcurl -->
<!-- note: default value is disabled. only enable if you want this! -->
<!-- <param name="ignore-cacert-check" value="true" /> -->
</settings>
</configuration>
conf/curl/autoload_configs/xml_curl.conf.xml 0 → 100644
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<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<!-- 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://freeswitch.org/fs_curl/index.php" bindings="dialplan|directory|configuration"/>
<!-- set this to provide authentication credentials to the server -->
<!--<param name="gateway-credentials" value="muser:mypass"/>-->
<!-- set to true to disable Expect: 100-continue lighttpd requires this setting -->
<!--<param name="disable-100-continue" value="true"/>-->
<!-- optional: if enabled this will disable CA root certificate checks by libcurl -->
<!-- note: default value is disabled. only enable if you want this! -->
<!-- <param name="ignore-cacert-check" value="true" /> -->
<!-- one or more of these imply you want to pick the exact variables that are transmitted -->
<!--<param name="enable-post-var" value="Unique-ID"/>-->
</binding>
</bindings>
</configuration>
conf/curl/autoload_configs/xml_rpc.conf.xml 0 → 100644
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<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>
conf/curl/autoload_configs/zeroconf.conf.xml 0 → 100644
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<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
<param name="publish" value="yes"/>
<param name="browse" value="_sip._udp"/>
</settings>
</configuration>
conf/curl/freeswitch.xml 0 → 100644
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<include>
<!--#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 compile the full xml document to ${prefix}/log/freeswitch.xml.fsxml
Don't modify it while freeswitch is running cos it is mem mapped in most cases =D
The same can be achieved with the <X-PRE-PROCESS> tag where the attrs 'cmd' and 'data' are
parsed in the same way.
-->
<X-PRE-PROCESS cmd="include" data="vars.xml"/>
<section name="configuration" description="Various Configuration">
<X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
</section>
<!-- mod_dingaling is reliant on the vcard data in the "directory" section. -->
<!-- mod_sofia is reliant on the user data for authorization -->
<section name="directory" description="User Directory">
<X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
</section>
<!-- phrases section (under development still) -->
<section name="phrases" description="Speech Phrase Management">
<macros>
<X-PRE-PROCESS cmd="include" data="lang/en/*.xml"/>
</macros>
</section>
</include>
conf/curl/mime.types 0 → 100644
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差异被折叠。
conf/insideout/autoload_configs/acl.conf.xml 0 → 100644
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<configuration name="acl.conf" description="Network Lists">
<network-lists>
<list name="dl-candidates" default="allow">
<node type="deny" cidr="10.0.0.0/8"/>
<node type="deny" cidr="172.16.0.0/12"/>
<node type="deny" cidr="192.168.0.0/16"/>
</list>
<list name="rfc1918" default="deny">
<node type="allow" cidr="10.0.0.0/8"/>
<node type="allow" cidr="172.16.0.0/12"/>
<node type="allow" cidr="192.168.0.0/16"/>
</list>
<list name="lan" default="allow">
<node type="deny" cidr="192.168.42.0/24"/>
<node type="allow" cidr="192.168.42.42/32"/>
</list>
<list name="strict" default="deny">
<node type="allow" cidr="208.102.123.124/32"/>
</list>
<!--
This will traverse the directory adding all users
with the cidr= tag to this ACL, when this ACL matches
the users variables and params apply as if they
digest authenticated.
-->
<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
</list>
</network-lists>
</configuration>
conf/insideout/autoload_configs/alsa.conf.xml 0 → 100644
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<configuration name="alsa.conf" description="Soundcard Endpoint">
<settings>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="N800 Alsa"/>
<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>
conf/insideout/autoload_configs/cdr_csv.conf.xml 0 → 100644
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<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<!-- 'cdr-csv' will always be appended to log-base -->
<!--<param name="log-base" value="/var/log"/>-->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
<param name="rotate-on-hup" value="true"/>
<!-- may be a b or ab -->
<param name="legs" value="a"/>
</settings>
<templates>
<template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
<template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
<template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
</templates>
</configuration>
conf/insideout/autoload_configs/conference.conf.xml 0 → 100644
浏览文件 @ f6514066
<!-- http://wiki.freeswitch.org/wiki/Mod_conference -->
<!-- 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 certain presence on startup . -->
<advertise>
<room name="3001@$${domain}" 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="$${domain}"/>
<!-- Sample Rate-->
<param name="rate" value="8000"/>
<!-- Number of milliseconds per frame -->
<param name="interval" value="20"/>
<!-- Energy level required for audio to be sent to the other users -->
<param name="energy-level" value="300"/>
<!--Can be | delim of waste|mute|deaf waste will always transmit data to each channel
even during silence -->
<!--<param name="member-flags" value="waste"/>-->
<!-- Name of the caller control group to use for this profile -->
<!-- <param name="caller-controls" value="some name"/> -->
<!-- TTS Engine to use -->
<!--<param name="tts-engine" value="cepstral"/>-->
<!-- TTS Voice to use -->
<!--<param name="tts-voice" value="david"/>-->
<!-- If TTS is enabled all audio-file params beginning with -->
<!-- 'say:' will be considered text to say with TTS -->
<!-- Set a default path here so you can use relative paths in the other sound params-->
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<!-- 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="conference/conf-muted.wav"/>
<!-- File to play to acknowledge unmuted -->
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<!-- File to play if you are alone in the conference -->
<param name="alone-sound" value="conference/conf-alone.wav"/>
<!-- File to play endlessly (nobody will ever be able to talk) -->
<!--<param name="perpetual-sound" value="perpetual.wav"/>-->
<!-- File to play when you're alone (music on hold)-->
<param name="moh-sound" value="$${hold_music}"/>
<!-- File to play when you join the conference -->
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<!-- File to play when you leave the conference -->
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<!-- File to play when you ae ejected from the conference -->
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<!-- File to play when the conference is locked -->
<param name="locked-sound" value="conference/conf-locked.wav"/>
<!-- File to play when the conference is locked during the call-->
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<!-- File to play when the conference is unlocked during the call-->
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<!-- File to play to prompt for a pin -->
<param name="pin-sound" value="conference/conf-pin.wav"/>
<!-- File to play to when the pin is invalid -->
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="comfort-noise-level" value="1400"/>
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<!-- Suppress start and stop talking events -->
<!-- <param name="suppress-events" value="start-talking,stop-talking"/> -->
<!-- enable comfort noise generation -->
<param name="comfort-noise" value="true"/>
<!-- Uncomment auto-record to toggle recording every conference call. -->
<!-- Another valid value is shout://user:pass@server.com/live.mp3 -->
<!--
<param name="auto-record" value="$${base_dir}/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
</profile>
<profile name="wideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!--<param name="tts-engine" value="flite"/>-->
<!--<param name="tts-voice" value="kal16"/>-->
</profile>
<profile name="ultrawideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="32000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
<!--<param name="tts-engine" value="flite"/>-->
<!--<param name="tts-voice" value="kal16"/>-->
</profile>
<!--
<profile name="cdquality">
<param name="domain" value="$${domain}"/>
<param name="rate" value="48000"/>
<param name="interval" value="10"/>
<param name="energy-level" value="300"/>
<param name="sound-prefix" value="$${base_dir}/sounds/en/us/callie"/>
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${hold_music}"/>
<param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
<param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/>
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<param name="locked-sound" value="conference/conf-locked.wav"/>
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<param name="pin-sound" value="conference/conf-pin.wav"/>
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<param name="caller-id-number" value="$${outbound_caller_id}"/>
<param name="comfort-noise" value="true"/>
</profile>
-->
</profiles>
</configuration>
conf/insideout/autoload_configs/console.conf.xml 0 → 100644
浏览文件 @ f6514066
<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>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
See examples below
The following map is the default, which is all debug levels enabled:
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
Example: the following turns on debugging for error and critical levels only
<map name="all" value="err,crit"/>
NOTE: using map name="all" will override any other settings! If you
want a more specific set of console messages then you will need
to specify which files and/or functions you want to have debug
messages. One option is to turn on just the more critical
messages with map name="all", then specify the other types of
console messages you want to see for various files and functions.
Example: turn on ERROR, CRIT, ALERT for all modules, then specify other
levels for various modules and functions
<map name="all" value="err,crit,alert"/>
<map name="switch_loadable_module_process" value="all"/>
<map name="mod_local_stream.c" value="warning,debug"/>
<map name="mod_sndfile.c" value="warning,info,debug"/>
-->
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
<!--
You can use or modify this sample set of mappings. It turns on higher
level messages for all modules and then specifies extra lower level
messages for OpenZAP, Sofia, and switch core messages.
<map name="all" value="warning,err,crit,alert"/>
<map name="zap_analog.c" value="all"/>
<map name="zap_io.c" value="all"/>
<map name="zap_isdn.c" value="all"/>
<map name="zap_zt.c" value="all"/>
<map name="mod_openzap" value="all"/>
<map name="sofia.c" value="notice"/>
<map name="switch_core_state_machine.c" value="all"/>
-->
</mappings>
<settings>
<!-- comment or set to false for no color logging -->
<param name="colorize" value="true"/>
<param name="loglevel" value="$${console_loglevel}"/>
</settings>
</configuration>
conf/insideout/autoload_configs/dialplan_directory.conf.xml 0 → 100644
浏览文件 @ f6514066
<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>
conf/insideout/autoload_configs/enum.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="enum.conf" description="ENUM Module">
<settings>
<param name="default-root" value="e164.org"/>
<param name="default-isn-root" value="freenum.org"/>
<param name="query-timeout" value="10"/>
<param name="auto-reload" value="true"/>
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1"/>
<!--<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>-->
<!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
</routes>
</configuration>
conf/insideout/autoload_configs/event_multicast.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="event_multicast.conf" description="Multicast Event">
<settings>
<param name="address" value="225.1.1.1"/>
<param name="port" value="4242"/>
<param name="bindings" value="all"/>
</settings>
</configuration>
conf/insideout/autoload_configs/event_socket.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="listen-ip" value="0.0.0.0"/>
<param name="listen-port" value="8021"/>
<param name="password" value="ClueCon"/>
<!--<param name="apply-inbound-acl" value="lan"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/fax.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="fax.conf" description="FAX application configuration">
<settings>
<param name="use-ecm" value="true"/>
<param name="verbose" value="false"/>
<param name="disable-v17" value="false"/>
<param name="ident" value="SpanDSP Fax Ident"/>
<param name="header" value="SpanDSP Fax Header"/>
<param name="spool-dir" value="/tmp"/>
<param name="file-prefix" value="faxrx"/>
</settings>
</configuration>
conf/insideout/autoload_configs/fifo.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="fifo.conf" description="FIFO Configuration">
<fifos>
<fifo name="cool_fifo@$${domain}" importance="0">
<!--<member timeout="60" simo="1" lag="20">user/1005@$${domain}</member>-->
</fifo>
</fifos>
</configuration>
conf/insideout/autoload_configs/iax.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="iax.conf" description="IAX Configuration">
<settings>
<param name="debug" value="0"/>
<param name="ip" value="$${local_ip_v4}"/>
<param name="port" value="4569"/>
<param name="context" value="public"/>
<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>
conf/insideout/autoload_configs/ivr.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="ivr.conf" description="IVR menus">
<menus>
<!-- demo IVR setup -->
<!-- demo IVR, Main Menu -->
<menu name="demo_ivr"
greet-long="phrase:demo_ivr_main_menu"
greet-short="phrase:demo_ivr_main_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
timeout ="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="4">
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/>
<entry action="menu-exec-app" digits="2" param="transfer 9996 XML default"/> <!-- FS echo -->
<entry action="menu-exec-app" digits="3" param="transfer 9999 XML default"/> <!-- MOH -->
<entry action="menu-sub" digits="4" param="demo_ivr_submenu"/> <!-- demo sub menu -->
<entry action="menu-exec-app" digits="5" param="transfer 1234*256 enum"/> <!-- Screaming monkeys -->
<entry action="menu-exec-app" digits="/^(10[01][0-9])$/" param="transfer $1 XML features"/>
<entry action="menu-top" digits="9"/> <!-- Repeat this menu -->
</menu>
<!-- Demo IVR, Sub Menu -->
<menu name="demo_ivr_submenu"
greet-long="phrase:demo_ivr_sub_menu"
greet-short="phrase:demo_ivr_sub_menu_short"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
timeout="15000"
max-failures="3"
max-timeouts="3">
<entry action="menu-top" digits="*"/>
</menu>
</menus>
</configuration>
<!-- TTS sample; non-functional but it demonstrates say: and TTS -->
<!--
<menu name="demo3"
greet-long="say:Press 1 to join the conference, Press 2 to join the other conference"
greet-short="say:Press 1 to join the conference, Press 2 to join the other conference"
invalid-sound="say:invalid extension"
exit-sound="say:exit sound"
timeout ="15000"
max-failures="3">
<entry action="menu-exit" digits="*"/>
<entry action="menu-playback" digits="1" param="say:You pressed 1"/>
<entry action="menu-call-transfer" digits="2" param=
"1000"/>
<entry action="menu-call-transfer" digits="3" param="1001"/>
</menu>
-->
conf/insideout/autoload_configs/java.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="java.conf" description="Java Plug-Ins">
<!-- Path to the Java 1.6 virtual machine to use -->
<javavm path="/usr/java/jdk1.6.0/jre/lib/i386/client/libjvm.so"/>
<!-- Options to pass to Java -->
<options>
<!-- Your class path (make sure freeswitch.jar is on it) -->
<option value="-Djava.class.path=$${base_dir}/scripts/freeswitch.jar"/>
<!-- Enable remote debugging -->
<option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000"/>
</options>
</configuration>
conf/insideout/autoload_configs/limit.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="limit.conf" description="LIMIT Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/local_stream.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="local_stream.conf" description="stream files from local dir">
<directory name="moh/8000" path="$${base_dir}/sounds/music/8000">
<param name="rate" value="8000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/16000" path="$${base_dir}/sounds/music/16000">
<param name="rate" value="16000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/32000" path="$${base_dir}/sounds/music/32000">
<param name="rate" value="32000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<!--
<directory name="moh/48000" path="$${base_dir}/sounds/music/48000">
<param name="rate" value="48000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="10"/>
<param name="timer-name" value="soft"/>
</directory>
-->
</configuration>
conf/insideout/autoload_configs/logfile.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="logfile.conf" description="File Logging">
<settings>
<!-- true to auto rotate on HUP, false to open/close -->
<param name="rotate-on-hup" value="true"/>
</settings>
<profiles>
<profile name="default">
<settings>
<!-- File to log to -->
<!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
<!-- At this length in bytes rotate the log file (0 for never) -->
<!--<param name="rollover" value="10485760"/>-->
</settings>
<mappings>
<!--
name can be a file name, function name or 'all'
value is one or more of debug,info,notice,warning,err,crit,alert,all
Please see comments in console.conf.xml for more information
-->
<map name="all" value="debug,info,notice,warning,err,crit,alert"/>
</mappings>
</profile>
</profiles>
</configuration>
conf/insideout/autoload_configs/lua.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="lua.conf" description="LUA Configuration">
<settings>
<!--
Specify local directories that will be searched for LUA modules
These entries will be pre-pended to the LUA_CPATH environment variable
-->
<!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
<!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> -->
<!--
Specify local directories that will be searched for LUA scripts
These entries will be pre-pended to the LUA_PATH environment variable
-->
<!-- <param name="script-directory" value="/usr/local/lua/?.lua"/> -->
<!-- <param name="script-directory" value="$${base_dir}/scripts/?.lua"/> -->
<!--<param name="xml-handler-script" value="/dp.lua"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a lua script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.lua"/>-->
<!--<param name="startup-script" value="startup_script_2.lua"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/modules.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<load module="mod_logfile"/>
<!-- <load module="mod_syslog"/> -->
<!--<load module="mod_yaml"/>-->
<!-- 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"/> -->
<!-- <load module="mod_xml_cdr"/> -->
<!-- Event Handlers -->
<load module="mod_cdr_csv"/>
<!-- <load module="mod_event_multicast"/> -->
<load module="mod_event_socket"/>
<!-- <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_alsa"/> -->
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_openzap"/> -->
<!-- Applications -->
<load module="mod_commands"/>
<load module="mod_conference"/>
<load module="mod_dptools"/>
<load module="mod_expr"/>
<load module="mod_fifo"/>
<load module="mod_voicemail"/>
<load module="mod_limit"/>
<load module="mod_esf"/>
<load module="mod_fsv"/>
<!-- SNOM Module -->
<!--<load module="mod_snom"/>-->
<!-- Dialplan Interfaces -->
<!-- <load module="mod_dialplan_directory"/> -->
<load module="mod_dialplan_xml"/>
<load module="mod_dialplan_asterisk"/>
<!-- Codec Interfaces -->
<load module="mod_voipcodecs"/>
<load module="mod_g723_1"/>
<load module="mod_g729"/>
<load module="mod_amr"/>
<load module="mod_ilbc"/>
<load module="mod_speex"/>
<load module="mod_h26x"/>
<!--<load module="mod_siren"/>-->
<!-- File Format Interfaces -->
<load module="mod_sndfile"/>
<load module="mod_native_file"/>
<!--For icecast/mp3 streams/files-->
<!--<load module="mod_shout"/>-->
<!--For local streams (play all the files in a directory)-->
<load module="mod_local_stream"/>
<load module="mod_tone_stream"/>
<!-- Timers -->
<!-- Languages -->
<load module="mod_spidermonkey"/>
<!-- <load module="mod_perl"/> -->
<!-- <load module="mod_python"/> -->
<!-- <load module="mod_java"/> -->
<load module="mod_lua"/>
<!-- ASR /TTS -->
<!-- <load module="mod_flite"/> -->
<!-- <load module="mod_pocketsphinx"/> -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_openmrcp"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Say -->
<load module="mod_say_en"/>
<!-- <load module="mod_say_zh"/> -->
</modules>
</configuration>
conf/insideout/autoload_configs/opal.conf.xml 0 → 100644
浏览文件 @ f6514066
<configuration name="opal.conf" description="Opal Endpoints">
<settings>
<param name="trace-level" value="4"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>
conf/insideout/autoload_configs/openzap.conf.xml 0 → 100644
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<configuration name="openzap.conf" description="OpenZAP Configuration">
<settings>
<param name="debug" value="0"/>
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
</settings>
<pri_spans>
<span name="PRI_1">
<!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="5ess"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
<span name="PRI_2">
<param name="q921loglevel" value="alert"/>
<param name="q931loglevel" value="alert"/>
<param name="mode" value="user"/>
<param name="dialect" value="5ess"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</pri_spans>
<!-- one entry here per openzap span -->
<analog_spans>
<span id="1">
<!--<param name="hold-music" value="$${moh_uri}"/>-->
<!--<param name="enable-analog-option" value="call-swap"/>-->
<!--<param name="enable-analog-option" value="3-way"/>-->
<param name="tonegroup" value="us"/>
<param name="digit-timeout" value="2000"/>
<param name="max-digits" value="11"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="enable-callerid" value="true"/>
<!-- regex to stop dialing when it matches -->
<!--<param name="dial-regex" value="5555"/>-->
<!-- regex to stop dialing when it does not match -->
<!--<param name="fail-dial-regex" value="^5"/>-->
</span>
</analog_spans>
</configuration>
conf/insideout/autoload_configs/perl.conf.xml 0 → 100644
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<configuration name="perl.conf" description="PERL Configuration">
<settings>
<!--<param name="xml-handler-script" value="/tmp/xml.pl"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a perl script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--param name="startup-script" value="startup_script_1.pl"/-->
<!--param name="startup-script" value="startup_script_2.pl"/-->
</settings>
</configuration>
conf/insideout/autoload_configs/pocketsphinx.conf.xml 0 → 100644
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<configuration name="pocketsphinx.conf" description="PocketSphinx ASR Configuration">
<settings>
<param name="threshold" value="400"/>
<param name="silence-hits" value="25"/>
<param name="listen-hits" value="1"/>
<param name="auto-reload" value="true"/>
<!--<param name="narrowband-model" value="communicator"/>-->
<!--<param name="wideband-model" value="wsj1"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/portaudio.conf.xml 0 → 100644
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<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="$${outbound_caller_name}"/>
<param name="cid-num" value="$${outbound_caller_id}"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="8000"/>
<param name="codec-ms" value="20"/>
</settings>
</configuration>
conf/insideout/autoload_configs/post_load_modules.conf.xml 0 → 100644
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<configuration name="post_load_modules.conf" description="Modules">
<modules>
</modules>
</configuration>
conf/insideout/autoload_configs/python.conf.xml 0 → 100644
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<configuration name="python.conf" description="PYTHON Configuration">
<settings>
<!--<param name="xml-handler-script" value="dp"/>-->
<!--<param name="xml-handler-bindings" value="dialplan"/>-->
<!--
The following options identifies a py script that is launched
at startup and may live forever in the background.
You can define multiple lines, one for each script you
need to run.
-->
<!--<param name="startup-script" value="startup_script_1.py"/>-->
<!--<param name="startup-script" value="startup_script_2.py"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/rss.conf.xml 0 → 100644
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<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>
conf/insideout/autoload_configs/shout.conf.xml 0 → 100644
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<configuration name="shout.conf" description="mod shout config">
<settings>
<!-- Don't change these unless you are insane -->
<!--<param name="decoder" value="i586"/>-->
<!--<param name="volume" value=".1"/>-->
<!--<param name="outscale" value="8192"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/sofia.conf.xml 0 → 100644
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<configuration name="sofia.conf" description="sofia Endpoint">
<global_settings>
<param name="log-level" value="0"/>
<!-- <param name="auto-restart" value="false"/> -->
<param name="debug-presence" value="0"/>
</global_settings>
<!--
The rabbit hole goes deep. This includes all the
profiles in the sip_profiles directory that is up
one level from this directory.
-->
<profiles>
<X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
</profiles>
</configuration>
conf/insideout/autoload_configs/spidermonkey.conf.xml 0 → 100644
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<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_socket"/>
<!--<load module="mod_spidermonkey_odbc"/>-->
</modules>
</configuration>
conf/insideout/autoload_configs/switch.conf.xml 0 → 100644
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<configuration name="switch.conf" description="Core Configuration">
<cli-keybindings>
<key name="1" value="$${function_key_1}"/>
<key name="2" value="$${function_key_2}"/>
<key name="3" value="$${function_key_3}"/>
<key name="4" value="$${function_key_4}"/>
<key name="5" value="$${function_key_5}"/>
<key name="6" value="$${function_key_6}"/>
<key name="7" value="$${function_key_7}"/>
<key name="8" value="$${function_key_8}"/>
<key name="9" value="$${function_key_9}"/>
<key name="10" value="$${function_key_10}"/>
<key name="11" value="$${function_key_11}"/>
<key name="12" value="$${function_key_12}"/>
</cli-keybindings>
<settings>
<!--Colorize the Console -->
<param name="colorize-console" value="true"/>
<!--Most channels to allow at once -->
<param name="max-sessions" value="1000"/>
<!--Most channels to create per second -->
<param name="sessions-per-second" value="30"/>
<!-- Default Global Log Level - value is one of debug,info,notice,warning,err,crit,alert -->
<param name="loglevel" value="debug"/>
<!--Try to catch any crashes that can be recoverable (in the context of a call)-->
<param name="crash-protection" value="false"/>
<!--<param name="max_dtmf_duration" value="192000"/>-->
<!--<param name="default_dtmf_duration" value="8000"/>-->
<!--
If you want to send out voicemail notifications via Windows you'll need to change the mailer-app
variable to the setting below:
<param name="mailer-app" value="msmtp"/>
Donot change mailer-app-args.
You will also need to download a sendmail clone for Windows (msmtp). This version works without issue:
http://msmtp.sourceforge.net/index.html. Download and copy the .exe to %winddir%\system32.
You'll need to create a small config file for smtp credentials (host name, authentication, tls, etc.) in
%USERPROFILE%\Application Data\ called "msmtprc.txt". Below is a sample copy of this file:
###################################
# The SMTP server of the provider.
account provider
host smtp.myisp.com
from john@myisp.com
auth login
user johndoe
password mypassword
# Set a default account
account default : provider
###################################
-->
<param name="mailer-app" value="sendmail"/>
<param name="mailer-app-args" value="-t"/>
<param name="dump-cores" value="yes"/>
<!--RTP port range -->
<!--<param name="rtp-start-port" value="16384"/>-->
<!--<param name="rtp-end-port" value="32768"/>-->
</settings>
</configuration>
conf/insideout/autoload_configs/syslog.conf.xml 0 → 100644
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<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="loglevel" value="warning"/>
</settings>
</configuration>
conf/insideout/autoload_configs/voicemail.conf.xml 0 → 100644
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<configuration name="voicemail.conf" description="Voicemail">
<settings>
</settings>
<profiles>
<profile name="default">
<param name="file-extension" value="wav"/>
<param name="terminator-key" value="#"/>
<param name="max-login-attempts" value="3"/>
<param name="digit-timeout" value="10000"/>
<param name="min-record-len" value="3"/>
<param name="max-record-len" value="300"/>
<param name="tone-spec" value="%(1000, 0, 640)"/>
<param name="callback-dialplan" value="XML"/>
<param name="callback-context" value="default"/>
<param name="play-new-messages-key" value="1"/>
<param name="play-saved-messages-key" value="2"/>
<param name="main-menu-key" value="0"/>
<param name="config-menu-key" value="5"/>
<param name="record-greeting-key" value="1"/>
<param name="choose-greeting-key" value="2"/>
<param name="change-pass-key" value="6"/>
<param name="record-name-key" value="3"/>
<param name="record-file-key" value="3"/>
<param name="listen-file-key" value="1"/>
<param name="save-file-key" value="2"/>
<param name="delete-file-key" value="7"/>
<param name="undelete-file-key" value="8"/>
<param name="email-key" value="4"/>
<param name="pause-key" value="0"/>
<param name="restart-key" value="1"/>
<param name="ff-key" value="6"/>
<param name="rew-key" value="4"/>
<param name="record-silence-threshold" value="200"/>
<param name="record-silence-hits" value="2"/>
<param name="web-template-file" value="web-vm.tpl"/>
<!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
<!--<param name="record-sample-rate" value="11025"/>-->
<!-- the next two both must be set for this to be enabled
the extension is in the format of <dest> [<dialplan>] [<context>]
-->
<param name="operator-extension" value="operator XML default"/>
<param name="operator-key" value="9"/>
<param name="vmain-extension" value="vmain XML default"/>
<param name="vmain-key" value="*"/>
<!-- playback created files as soon as they were recorded by default -->
<!--<param name="auto-playback-recordings" value="true"/>-->
<email>
<param name="template-file" value="voicemail.tpl"/>
<param name="notify-template-file" value="notify-voicemail.tpl"/>
<!-- this is the format voicemail_time will have -->
<param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
<param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
</email>
<!--<param name="storage-dir" value="/tmp"/>-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="record-comment" value="Your Comment"/>-->
<!--<param name="record-title" value="Your Title"/>-->
<!--<param name="record-copyright" value="Your Copyright"/>-->
</profile>
</profiles>
</configuration>
conf/insideout/autoload_configs/xml_cdr.conf.xml 0 → 100644
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<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
<settings>
<!-- the url to post to if blank web posting is disabled -->
<!-- <param name="url" value="http://localhost/cdr_curl/post.php"/> -->
<!-- optional: credentials to send to web server -->
<!-- <param name="cred" value="user:pass"/> -->
<!-- the total number of retries (not counting the first 'try') to post to webserver incase of failure -->
<!-- <param name="retries" value="2"/> -->
<!-- delay between retries in seconds, default is 5 seconds -->
<!-- <param name="delay" value="1"/> -->
<!-- optional: if not present we do not log every record to disk -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank value will default to ${prefix}/logs/xml_cdr -->
<param name="log-dir" value=""/>
<!-- optional: if not present we do log the b leg -->
<!-- true or false if we should create a cdr for the b leg of a call-->
<param name="log-b-leg" value="false"/>
<!-- optional: if not present, all filenames are the uuid of the call -->
<!-- true or false if a leg files are prefixed "a_" -->
<param name="prefix-a-leg" value="true"/>
<!-- encode the post data may be 'true' for url encoding, 'false' for no encoding or 'base64' for base64 encoding -->
<param name="encode" value="true"/>
<!-- optional: set to true to disable Expect: 100-continue lighttpd requires this setting -->
<!--<param name="disable-100-continue" value="true"/>-->
<!-- optional: full path to the error log dir for failed web posts if not specified its the same as log-dir -->
<!-- either an absolute path, a relative path assuming ${prefix}/logs or a blank or omitted value will default to ${prefix}/logs/xml_cdr -->
<!-- <param name="err-log-dir" value="/tmp"/> -->
<!-- optional: if enabled this will disable CA root certificate checks by libcurl -->
<!-- note: default value is disabled. only enable if you want this! -->
<!-- <param name="ignore-cacert-check" value="true" /> -->
</settings>
</configuration>
conf/insideout/autoload_configs/xml_curl.conf.xml 0 → 100644
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<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<!-- 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.freeswitch.org/gateway.xml" bindings="dialplan"/> -->
<!-- set this to provide authentication credentials to the server -->
<!--<param name="gateway-credentials" value="muser:mypass"/>-->
<!-- set to true to disable Expect: 100-continue lighttpd requires this setting -->
<!--<param name="disable-100-continue" value="true"/>-->
<!-- optional: if enabled this will disable CA root certificate checks by libcurl -->
<!-- note: default value is disabled. only enable if you want this! -->
<!-- <param name="ignore-cacert-check" value="true" /> -->
<!-- one or more of these imply you want to pick the exact variables that are transmitted -->
<!--<param name="enable-post-var" value="Unique-ID"/>-->
</binding>
</bindings>
</configuration>
conf/insideout/autoload_configs/xml_rpc.conf.xml 0 → 100644
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<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>
conf/insideout/autoload_configs/zeroconf.conf.xml 0 → 100644
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<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
<param name="publish" value="yes"/>
<param name="browse" value="_sip._udp"/>
</settings>
</configuration>
conf/insideout/dialplan/default.xml 0 → 100644
浏览文件 @ f6514066
差异被折叠。
conf/insideout/dialplan/default/00_pizza_demo.xml 0 → 100644
浏览文件 @ f6514066
<include>
<extension name="pizza_demo">
<condition field="destination_number" expression="^(pizza|74992)$"/>
<condition field="${module_exists(mod_spidermonkey)}" expression="true"/>
<condition field="${module_exists(mod_pocketsphinx)}" expression="true">
<action application="javascript" data="ps_pizza.js"/>
</condition>
</extension>
</include>
conf/insideout/dialplan/default/01_example.com.xml 0 → 100644
浏览文件 @ f6514066
<include>
<extension name="local.example.com">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{7})$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/>
</condition>
</extension>
<extension name="domestic.example.com">
<condition field="${toll_allow}" expression="domestic"/>
<condition field="destination_number" expression="^(\d{11})$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
</condition>
</extension>
<extension name="international.example.com">
<condition field="${toll_allow}" expression="international"/>
<condition field="destination_number" expression="^(011\d+)$">
<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
<action application="bridge" data="sofia/gateway/${default_gateway}/$1"/>
</condition>
</extension>
</include>
conf/insideout/dialplan/default/02_conferences.xml 0 → 100644
浏览文件 @ f6514066
<include>
<extension name="international.example.com">
<condition field="destination_number" expression="^(42|888|899)$">
<action application="set" data="effective_caller_id_number=42"/>
<action application="set" data="effective_caller_id_name=Ray"/>
<action application="bridge" data="sofia/external/$1@conference.freeswitch.org"/>
</condition>
</extension>
</include>
conf/insideout/dialplan/default/99999_enum.xml 0 → 100644
浏览文件 @ f6514066
<include>
<extension name="enum">
<condition field="${module_exists(mod_enum)}" expression="true"/>
<condition field="destination_number" expression="^(.*)$">
<action application="transfer" data="$1 enum"/>
</condition>
</extension>
</include>
conf/insideout/dialplan/features.xml 0 → 100644
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<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="features">
<!-- In call Transfer for phones without a transfer button -->
<extension name="dx">
<condition field="destination_number" expression="^dx$">
<action application="answer"/>
<action application="read" data="11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
<action application="execute_extension" data="is_transfer XML features"/>
</condition>
</extension>
<extension name="is_transfer">
<condition field="destination_number" expression="^is_transfer$"/>
<condition field="${digits}" expression="^(\d+)$">
<action application="transfer" data="-bleg ${digits} XML default"/>
<anti-action application="eval" data="cancel transfer"/>
</condition>
</extension>
<!-- Used to transfer both legs into a conference -->
<extension name="cf">
<condition field="destination_number" expression="^cf$">
<action application="answer"/>
<action application="transfer" data="-both 30${dialed_extension:2} XML default"/>
</condition>
</extension>
<extension name="please_hold">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="answer"/>
<action application="sleep" data="1500"/>
<action application="playback" data="ivr/ivr-hold_connect_call.wav"/>
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<extension name="is_secure">
<!-- Only Truly consider it secure if its TLS and SRTP -->
<condition field="${sip_via_protocol}" expression="tls"/>
<condition field="${sip_secure_media_confirmed}" expression="^true$">
<action application="sleep" data="1000"/>
<action application="playback" data="misc/call_secured.wav"/>
<anti-action application="eval" data="not_secure"/>
</condition>
</extension>
</context>
</include>
conf/insideout/dialplan/public.xml 0 → 100644
浏览文件 @ f6514066
<!--
NOTICE:
This context is usually accessed via the external sip profile sitting on port 5080.
It is recommended to have separate inbound and outbound contexts. Not only for security
but clearing up why you would need to do such a thing. You don't want outside un-authenticated
callers hitting your default context which allows dialing calls thru your providers and results
in Toll Fraud.
-->
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="public">
<extension name="unloop">
<condition field="$${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!--
Tag anything pass thru here as an outside_call so you can make sure not
to create any routing loops based on the conditions that it came from
the outside of the switch.
-->
<extension name="outside_call" continue="true">
<condition>
<action application="set" data="outside_call=true"/>
</condition>
</extension>
<extension name="call_debug" continue="true">
<condition field="$${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
</extension>
<!-- Try to get domain_name from the sip_req_params variable -->
<extension name="set_domain" continue="true">
<condition field="${domain_name}" expression="^$"/>
<condition field="source" expression="mod_sofia"/>
<condition field="${sip_req_params}" expression="domain_name=([A-Z-a-z0-9.]+)">
<!-- We need to export this so the B-Leg will have it after transfer too. -->
<action application="export" data="domain_name=$1"/>
<anti-action application="export" data="domain_name=${sip_req_host}"/>
</condition>
</extension>
<extension name="public_extensions">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
<!--
You can place files in the public directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="public/*.xml"/>
<!--
If you have made it this far lets challenge the caller and if they authenticate
lets try what they dialed in the default context. (commented out by default)
-->
<!--
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>
<extension name="transfer_to_default">
<condition>
<action application="transfer" data="${destination_number} XML default"/>
</condition>
</extension>
-->
</context>
</include>
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<include>
<extension name="public_did">
<condition field="destination_number" expression="^(5551212)$">
<!--
If you're hosting multiple domains you will want to set the
target_domain on these calls so they hit the proper domain after you
transfer the caller into the default context.
$${domain} is the default domain set from vars.xml but you can set it
to any domain you have setup in your user directory.
-->
<action application="set" data="domain_name=$${domain}"/>
<!-- This example maps the DID 5551212 to ring 1000 in the default context -->
<action application="transfer" data="1000 XML default"/>
</condition>
</extension>
</include>
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<!--
NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
FreeSWITCH works off the concept of users and domains just like email.
You have users that are in domains for example 1000@domain.com.
When freeswitch gets a register packet it looks for the user in the directory
based on the from or to domain in the packet depending on how your sofia profile
is configured. Out of the box the default domain will be the IP address of the
machine running FreeSWITCH. This IP can be found by typing "sofia status" at the
CLI. You will register your phones to the IP and not the hostname by default.
If you wish to register using the domain please open vars.xml in the root conf
directory and set the default domain to the hostname you desire. Then you would
use the domain name in the client instead of the IP address to register
with FreeSWITCH.
NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
-->
<include>
<!--the domain or ip (the right hand side of the @ in the addr-->
<domain name="$${domain}">
<params>
<param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
</params>
<variables>
<variable name="record_stereo" value="true"/>
<variable name="default_gateway" value="$${default_provider}"/>
<variable name="default_areacode" value="$${default_areacode}"/>
<variable name="transfer_fallback_extension" value="operator"/>
<variable name="use_profile" value="external"/>
</variables>
<X-PRE-PROCESS cmd="include" data="default/*.xml"/>
</domain>
</include>
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<include>
<user id="1000" mailbox="1000">
<params>
<param name="password" value="1234"/>
<param name="vm-password" value="1000"/>
</params>
<variables>
<variable name="toll_allow" value="domestic,international,local"/>
<variable name="accountcode" value="1000"/>
<variable name="user_context" value="default"/>
<variable name="effective_caller_id_name" value="Extension 1000"/>
<variable name="effective_caller_id_number" value="1000"/>
<variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
<variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
<variable name="callgroup" value="techsupport"/>
</variables>
</user>
</include>
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