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freeswitch.xml.fsxml
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<!--
NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
This is the FreeSWITCH default config. Everything you see before you now traverses
down into all the directories including files which include more files. The default
config comes out of the box already working in most situations as a PBX. This will
allow you to get started testing and playing with various things in FreeSWITCH.
Before you start to modify this default please visit this wiki page:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out.21
If all else fails you can read our FAQ located at:
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ
NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
-->
<document type="freeswitch/xml">
<!-- Preprocessor Variables
These are introduced when configuration strings must be consistent across modules.
WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
YOU SHOULD CHANGE THIS default_password value if you don't want to be subject to any
toll fraud in the future. It's your responsibility to secure your own system.
This default config is used to demonstrate the feature set of FreeSWITCH.
WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING
-->
<!-- Did you change it yet? -->
<!--
This setting is what sets the default domain FreeSWITCH will use if all else fails.
FreeSWICH will default to 10.5.5.208 unless changed. Changing this setting does
affect the sip authentication. Please review conf/directory/default.xml for more
information on this topic.
-->
<!--IP Adress-->
<!--
Enable ZRTP globally you can override this on a per channel basis
http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
-->
<!--
Examples of codec options: (module must be compiled and loaded)
codecname[@8000h|16000h|32000h[@XXi]]
XX is the frame size must be multples allowed for the codec
FreeSWITCH can support 10-120ms on some codecs.
We do not support exceeding the MTU of the RTP packet.
iLBC@30i - iLBC using mode=30 which will win in all cases.
DVI4@8000h@20i - IMA ADPCM 8kHz using 20ms ptime. (multiples of 10)
DVI4@16000h@40i - IMA ADPCM 16kHz using 40ms ptime. (multiples of 10)
speex@8000h@20i - Speex 8kHz using 20ms ptime.
speex@16000h@20i - Speex 16kHz using 20ms ptime.
speex@32000h@20i - Speex 32kHz using 20ms ptime.
BV16 - BroadVoice 16kb/s narrowband, 8kHz
BV32 - BroadVoice 32kb/s wideband, 16kHz
G7221@16000h - G722.1 16kHz (aka Siren 7)
G7221@32000h - G722.1C 32kHz (aka Siren 14)
CELT@32000h - CELT 32kHz, only 10ms supported
CELT@48000h - CELT 48kHz, only 10ms supported
GSM@40i - GSM 8kHz using 40ms ptime. (GSM is done in multiples of 20, Default is 20ms)
G722 - G722 16kHz using default 20ms ptime. (multiples of 10)
PCMU - G711 8kHz ulaw using default 20ms ptime. (multiples of 10)
PCMA - G711 8kHz alaw using default 20ms ptime. (multiples of 10)
G726-16 - G726 16kbit adpcm using default 20ms ptime. (multiples of 10)
G726-24 - G726 24kbit adpcm using default 20ms ptime. (multiples of 10)
G726-32 - G726 32kbit adpcm using default 20ms ptime. (multiples of 10)
G726-40 - G726 40kbit adpcm using default 20ms ptime. (multiples of 10)
AAL2-G726-16 - Same as G726-16 but using AAL2 packing. (multiples of 10)
AAL2-G726-24 - Same as G726-24 but using AAL2 packing. (multiples of 10)
AAL2-G726-32 - Same as G726-32 but using AAL2 packing. (multiples of 10)
AAL2-G726-40 - Same as G726-40 but using AAL2 packing. (multiples of 10)
LPC - LPC10 using 90ms ptime (only supports 90ms at this time in FreeSWITCH)
L16 - L16 isn't recommended for VoIP but you can do it. L16 can exceed the MTU rather quickly.
These are the passthru audio codecs:
G729 - G729 in passthru mode. (mod_g729)
G723 - G723.1 in passthru mode. (mod_g723_1)
AMR - AMR in passthru mode. (mod_amr)
These are the passthru video codecs: (mod_h26x)
H261 - H.261 Video
H263 - H.263 Video
H263-1998 - H.263-1998 Video
H263-2000 - H.263-2000 Video
H264 - H.264 Video
RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for.
96 - AMR
97 - iLBC (30)
98 - iLBC (20)
99 - Speex 8kHz, 16kHz, 32kHz
100 -
101 - telephone-event
102 -
103 -
104 -
105 -
106 - BV16
107 - G722.1 (16kHz)
108 -
109 -
110 -
111 -
112 -
113 -
114 - CELT 32kHz, 48kHz
115 - G722.1C (32kHz)
116 -
117 - SILK 8kHz
118 - SILK 12kHz
119 - SILK 16kHz
120 - SILK 24kHz
121 - AAL2-G726-40 && G726-40
122 - AAL2-G726-32 && G726-32
123 - AAL2-G726-24 && G726-24
124 - AAL2-G726-16 && G726-16
125 -
126 -
127 - BV32
-->
<!--
xmpp_client_profile and xmpp_server_profile
xmpp_client_profile can be any string.
xmpp_server_profile is appended to "dingaling_" to form the database name
containing the "subscriptions" table.
used by: dingaling.conf.xml enum.conf.xml
-->
<!--
THIS IS ONLY USED FOR DINGALING
bind_server_ip
Can be an ip address, a dns name, or "auto".
This determines an ip address available on this host to bind.
If you are separating RTP and SIP traffic, you will want to have
use different addresses where this variable appears.
Used by: dingaling.conf.xml
-->
<!-- NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
If you're going to load test FreeSWITCH please input real IP addresses
for external_rtp_ip and external_sip_ip
-->
<!-- external_rtp_ip
Can be an one of:
ip address: "12.34.56.78"
a stun server lookup: "stun:stun.server.com"
a DNS name: "host:host.server.com"
where fs.mydomain.com is a DNS A record-useful when fs is on
a dynamic IP address, and uses a dynamic DNS updater.
If unspecified, the bind_server_ip value is used.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an one of:
ip address: "12.34.56.78"
a stun server lookup: "stun:stun.server.com"
a DNS name: "host:host.server.com"
where fs.mydomain.com is a DNS A record-useful when fs is on
a dynamic IP address, and uses a dynamic DNS updater.
If unspecified, the bind_server_ip value is used.
Used by: sofia.conf.xml dingaling.conf.xml
-->
<!-- unroll-loops
Used to turn on sip loopback unrolling.
-->
<!-- outbound_caller_id and outbound_caller_name
The caller ID telephone number we should use when calling out.
Used by: conference.conf.xml and user directory for default
outbound callerid name and number.
-->
<!-- various debug and defaults -->
<!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
if true, the destination number is not included -->
<!--
Setting up your default sip provider is easy.
Below are some values that should work in most cases.
These are for conf/directory/default/example.com.xml
-->
<!-- true or false -->
<!--
SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
-->
<!-- Internal SIP Profile -->
<!-- External SIP Profile -->
<section name="configuration" description="Various Configuration">
<configuration name="abstraction.conf" description="Abstraction">
<apis>
<api name="user_name" description="Return Name for extension" syntax="<exten>" parse="(.*)" destination="user_data" argument="$1@default var effective_caller_id_name"/>
</apis>
</configuration>
<configuration name="acl.conf" description="Network Lists">
<network-lists>
<!--
These ACL's are automatically created on startup.
rfc1918.auto - RFC1918 Space
nat.auto - RFC1918 Excluding your local lan.
localnet.auto - ACL for your local lan.
loopback.auto - ACL for your local lan.
-->
<list name="lan" default="allow">
<node type="deny" cidr="192.168.42.0/24"/>
<node type="allow" cidr="192.168.42.42/32"/>
</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">
<!-- domain= is special it scans the domain from the directory to build the ACL -->
<node type="allow" domain="10.5.5.208"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
</list>
</network-lists>
</configuration>
<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>
<configuration name="mod_blacklist.conf" description="Blacklist module">
<lists>
<!--
Example blacklist, the referenced file contains blacklisted items, one entry per line
NOTE: make sure the file exists and is readable by FreeSWITCH.
<list name="example" filename="/usr/local/freeswitch/conf/blacklists/example.list"/>
-->
</lists>
</configuration>
<configuration name="callcenter.conf" description="CallCenter">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="dbname" value="/dev/shm/callcenter.db"/>-->
</settings>
<queues>
<queue name="support@default">
<param name="strategy" value="longest-idle-agent"/>
<param name="moh-sound" value="local_stream://moh"/>
<!--<param name="record-template" value="/home/kwyrozumski/work/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}.${destination_number}.${caller_id_number}.${uuid}.wav"/>-->
<param name="time-base-score" value="system"/>
<param name="max-wait-time" value="0"/>
<param name="max-wait-time-with-no-agent" value="0"/>
<param name="max-wait-time-with-no-agent-time-reached" value="5"/>
<param name="tier-rules-apply" value="false"/>
<param name="tier-rule-wait-second" value="300"/>
<param name="tier-rule-wait-multiply-level" value="true"/>
<param name="tier-rule-no-agent-no-wait" value="false"/>
<param name="discard-abandoned-after" value="60"/>
<param name="abandoned-resume-allowed" value="false"/>
</queue>
</queues>
<!-- WARNING: Configuration of XML Agents will be updated into the DB upon restart. -->
<!-- WARNING: Configuration of XML Tiers will reset the level and position if those were supplied. -->
<!-- WARNING: Agents and Tiers XML config shouldn't be used in a multi FS shared DB setup (Not currently supported anyway) -->
<agents>
<!--<agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000@default" status="Available" max-no-answer="3" wrap-up-time="10" reject-delay-time="10" busy-delay-time="60" />-->
</agents>
<tiers>
<!-- If no level or position is provided, they will default to 1. You should do this to keep db value on restart. -->
<!-- <tier agent="1000@default" queue="support@default" level="1" position="1"/> -->
</tiers>
</configuration>
<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<!-- 'cdr-csv' will always be appended to log-base -->
<!--<param name="log-base" value="/var/log"/>-->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
<param name="rotate-on-hup" value="true"/>
<!-- may be a b or ab -->
<param name="legs" value="a"/>
<!-- Only log in Master.csv -->
<!-- <param name="master-file-only" value="true"/> -->
</settings>
<templates>
<template name="sql">INSERT INTO cdr VALUES ("${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>
<configuration name="cdr_mongodb.conf" description="MongoDB CDR logger">
<settings>
<!-- Hostnames and IPv6 addrs not supported (yet) -->
<param name="host" value="127.0.0.1"/>
<param name="port" value="27017"/>
<!-- Namespace format is database.collection -->
<param name="namespace" value="test.cdr"/>
<!-- If true, create CDR for B-leg of call (default: true) -->
<param name="log-b-leg" value="false"/>
</settings>
</configuration>
<configuration name="cdr_pg_csv.conf" description="CDR PG CSV Format">
<settings>
<!-- See parameters for PQconnectdb() at http://www.postgresql.org/docs/8.4/static/libpq-connect.html -->
<param name="db-info" value="host=localhost dbname=cdr connect_timeout=10" />
<!-- CDR table name -->
<!--<param name="db-table" value="cdr"/>-->
<!-- Log a-leg (a), b-leg (b) or both (ab) -->
<param name="legs" value="a"/>
<!-- Directory in which to spool failed SQL inserts -->
<!-- <param name="spool-dir" value="/home/kwyrozumski/work/freeswitch/log/cdr-pg-csv"/> -->
<!-- Disk spool format if DB connection/insert fails - csv (default) or sql -->
<param name="spool-format" value="csv"/>
<param name="rotate-on-hup" value="true"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
</settings>
<schema>
<field var="local_ip_v4"/>
<field var="caller_id_name"/>
<field var="caller_id_number"/>
<field var="destination_number"/>
<field var="context"/>
<field var="start_stamp"/>
<field var="answer_stamp"/>
<field var="end_stamp"/>
<field var="duration" quote="false"/>
<field var="billsec" quote="false"/>
<field var="hangup_cause"/>
<field var="uuid"/>
<field var="bleg_uuid"/>
<field var="accountcode"/>
<field var="read_codec"/>
<field var="write_codec"/>
<!-- <field var="sip_hangup_disposition"/> -->
<!-- <field var="ani"/> -->
</schema>
</configuration>
<configuration name="cdr_sqlite.conf" description="SQLite CDR">
<settings>
<!-- SQLite database name (.db suffix will be automatically appended) -->
<!-- <param name="db-name" value="cdr"/> -->
<!-- CDR table name -->
<!-- <param name="db-table" value="cdr"/> -->
<!-- Log a-leg (a), b-leg (b) or both (ab) -->
<param name="legs" value="a"/>
<!-- Default template to use when inserting records -->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
</settings>
<templates>
<!-- Note that field order must match SQL table schema, otherwise insert will fail -->
<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}","${accountcode}"</template>
</templates>
</configuration>
<configuration name="cepstral.conf" description="Cepstral TTS configuration">
<settings>
<!--
Possible encodings:
* utf-8
* us-ascii
* iso8859-1 (default)
* iso8859-15
-->
<param name="encoding" value="utf-8"/>
</settings>
</configuration><configuration name="cidlookup.conf" description="cidlookup Configuration">
<settings>
<!-- comment out url to not setup a url based lookup -->
<param name="url" value="http://query.voipcnam.com/query.php?api_key=MYAPIKEY&number=${caller_id_number}"/>
<!-- comment out whitepages-apikey to not use whitepages.com, you must
get an API key from http://developer.whitepages.com/ -->
<param name="whitepages-apikey" value="MYAPIKEY"/>
<!-- set to false to not cache (in memcache) results from the url query -->
<param name="cache" value="true"/>
<!-- expire is in seconds -->
<param name="cache-expire" value="86400"/>
<param name="odbc-dsn" value="phone:phone:phone"/>
<!-- comment out sql to not setup a database (directory) lookup -->
<param name="sql" value="
SELECT name||' ('||type||')' AS name
FROM phonebook p JOIN numbers n ON p.id = n.phonebook_id
WHERE n.number='${caller_id_number}'
LIMIT 1
"/>
<!-- comment out citystate-sql to not setup a database (city/state)
lookup -->
<param name="citystate-sql" value="
SELECT ratecenter||' '||state as name
FROM npa_nxx_company_ocn
WHERE npa = ${caller_id_number:1:3} AND nxx = ${caller_id_number:4:3}
LIMIT 1
"/>
</settings>
</configuration>
<!-- 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="[email protected]" 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. Disabled if dist-dtmf member flag is set. -->
<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">
<!-- Directory to drop CDR's
'auto' means $PREFIX/logs/conference_cdr/<confernece_uuid>.cdr.xml
a non-absolute path means $PREFIX/logs/<value>/<confernece_uuid>.cdr.xml
absolute path means <value>/<confernece_uuid>.cdr.xml
-->
<!-- <param name="cdr-log-dir" value="auto"/> -->
<!-- Domain (for presence) -->
<param name="domain" value="10.5.5.208"/>
<!-- 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|dist-dtmf waste will always transmit data to each channel
even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls
via dtmf will be disabled. -->
<!--<param name="member-flags" value="waste"/>-->
<!-- Name of the caller control group to use for this profile -->
<!-- <param name="caller-controls" value="some name"/> -->
<!-- Name of the caller control group to use for the moderator in this profile -->
<!-- <param name="moderator-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 -->
<!-- Override the default path here, after which you use relative paths in the other sound params -->
<!-- Note: The default path is the conference's first caller's sound_prefix -->
<!--<param name="sound-prefix" value="/home/kwyrozumski/work/freeswitch/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="local_stream://moh"/>
<!-- 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"/>
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!--<param name="moderator-pin" value="54321"/>-->
<!-- Max number of times the user can be prompted for PIN -->
<!--<param name="pin-retries" value="3"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="FreeSWITCH"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="3340"/>
<!-- 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:[email protected]/live.mp3 -->
<!--
<param name="auto-record" value="/home/kwyrozumski/work/freeswitch/recordings/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
<!-- IVR digit machine timeouts -->
<!-- How much to wait between DTMF digits to match caller-controls -->
<!-- <param name="ivr-dtmf-timeout" value="500"/> -->
<!-- How much to wait for the first DTMF, 0 forever -->
<!-- <param name="ivr-input-timeout" value="0" /> -->
<!-- Delay before a conference is asked to be terminated -->
<!-- <param name="endconf-grace-time" value="120" /> -->
</profile>
<profile name="wideband">
<param name="domain" value="10.5.5.208"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<!--<param name="sound-prefix" value="/home/kwyrozumski/work/freeswitch/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="local_stream://moh"/>
<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="FreeSWITCH"/>
<param name="caller-id-number" value="3340"/>
<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="10.5.5.208"/>
<param name="rate" value="32000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<!--<param name="sound-prefix" value="/home/kwyrozumski/work/freeswitch/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="local_stream://moh"/>
<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="FreeSWITCH"/>
<param name="caller-id-number" value="3340"/>
<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="10.5.5.208"/>
<param name="rate" value="48000"/>
<param name="interval" value="10"/>
<param name="energy-level" value="300"/>
<!--<param name="sound-prefix" value="/home/kwyrozumski/work/freeswitch/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="local_stream://moh"/>
<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="FreeSWITCH"/>
<param name="caller-id-number" value="3340"/>
<param name="comfort-noise" value="true"/>
</profile>
<profile name="sla">
<param name="domain" value="10.5.5.208"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="caller-controls" value="none"/>
<param name="energy-level" value="200"/>
<param name="moh-sound" value="silence"/>
<param name="comfort-noise" value="true"/>
</profile>
</profiles>
</configuration>
<configuration name="console.conf" description="Console Logger">
<!-- pick a file name, a function name or 'all' -->
<!-- map as many as you need for specific debugging -->
<mappings>
<!--
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="info"/>
</settings>
</configuration>
<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>
<configuration name="dialplan_directory.conf" description="Dialplan Directory">
<settings>
<param name="directory-name" value="ldap"/>
<param name="host" value="ldap.freeswitch.org"/>
<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
<param name="pass" value="test"/>
<param name="base" value="dc=freeswitch,dc=org"/>
</settings>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="PCMU"/>
</settings>
<!-- Client Profile (Original mode) -->
<!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
<x-profile type="client">
<param name="name" value="xmppc"/>
<param name="login" value="[email protected]/talk"/>
<param name="password" value="mypass"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="message" value="Jingle all the way"/>
<param name="rtp-ip" value="auto"/>
<!-- <param name="ext-rtp-ip" value="auto-nat"/> -->
<param name="auto-login" value="true"/>
<!-- SASL "plain" or "md5" -->
<param name="sasl" value="plain"/>
<!-- if the server where the jabber is hosted is not the same as the one in the jid -->
<!--<param name="server" value="alternate.server.com"/>-->
<!-- Enable TLS or not -->
<param name="tls" value="true"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- default extension (if one cannot be determined) -->
<param name="exten" value="888"/>
<!-- VAD choose one -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!--<param name="vad" value="both"/>-->
<!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
<!--<param name="candidate-acl" value="wan.auto"/>-->
<param name="local-network-acl" value="localnet.auto"/>
</x-profile>
<!-- Component (Server to Server Login) -->
<!-- to use this profile take the x- away from the open and close tags so its <profile> and </profile> -->
<x-profile type="component">
<param name="name" value="xmpps"/>
<param name="password" value="secret"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="rtp-ip" value="auto"/>
<param name="server" value="jabber.server.org:5347"/>
<!-- disable to trade async for more calls -->
<param name="use-rtp-timer" value="true"/>
<!-- "_auto_" means the extension will be automaticly set to the called jid -->
<param name="exten" value="_auto_"/>
<!--<param name="vad" value="both"/>-->
<!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="candidate-acl" value="wan.auto"/>-->
</x-profile>
</configuration>
<configuration name="directory.conf" description="Directory">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--<param name="dbname" value="directory"/>-->
</settings>
<profiles>
<profile name="default">
<param name="max-menu-attempts" value="3"/>
<param name="min-search-digits" value="3"/>
<param name="terminator-key" value="#"/>
<param name="digit-timeout" value="3000"/>
<param name="max-result" value="5"/>
<param name="next-key" value="6"/>
<param name="prev-key" value="4"/>
<param name="switch-order-key" value="*"/>
<param name="select-name-key" value="1"/>
<param name="new-search-key" value="3"/>
<param name="search-order" value="last_name"/>
</profile>
</profiles>
</configuration>
<configuration name="distributor.conf" description="Distributor Configuration">
<lists>
<!-- every 10 calls to test you will get foo1 once and foo2 9 times...yes NINE TIMES! -->
<!-- this is not the same as 100 with 10 and 90 that would do foo1 10 times in a row then foo2 90 times in a row -->
<list name="test" total-weight="10">
<node name="foo1" weight="1"/>
<node name="foo2" weight="9"/>
</list>
</lists>
</configuration>
<configuration name="easyroute.conf" description="EasyRoute Module">
<settings>
<!-- These are kind Obvious -->
<param name="db-username" value="root"/>
<param name="db-password" value="password"/>
<param name="db-dsn" value="easyroute"/>
<!-- Default Technology and profile -->
<param name="default-techprofile" value="sofia/default"/>
<!-- IP or Hostname of Default Route -->
<param name="default-gateway" value="192.168.66.6"/>
<!-- Number of times to retry ODBC connection on connection problems, default is 120 -->
<param name="odbc-retries" value="120"/>
<!-- Customer Query. Use this with Care!!! We are not responsible if you mess
This up!!! Query *MUST* return columns in the following order!
gateway varchar(128) - contains destination gateway host:port pair (ex: 192.168.1.1:5060 )
group varchar(128) - contains optional group name
call_limit varchar(16) - contains optional call limit
tech_prefix varchar(128) - tech prefix used to build dial string (ex: sofia/default )
acctcode varchar(128) - used to set channel variable acctcode for logging into the CDRs
destination_number varchar(16) - Number returning for the query for building the dial string. (ex: 18005551212)
See Documentation on the Wiki for further information -->
<!-- <param name="custom-query" value="call FS_GET_SIP_LOCATION(%s);"/> -->
</settings>
</configuration>
<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+XMPP" regex="XMPP:(.*)" replace="dingaling/xmpps/$1"/>-->
</routes>
</configuration>
<configuration name="erlang_event.conf" description="Erlang Socket Client">
<settings>
<param name="listen-ip" value="0.0.0.0"/>
<param name="listen-port" value="8031"/>
<!-- Specify the first part of the node name
(the host part after the @ will be autodetected)
OR pass a complete nodename to avoid autodetection
eg. freeswitch@example or [email protected].
If you pass a complete node name, the 'shortname' parameter has no effect. -->
<param name="nodename" value="freeswitch"/>
<!-- Specify this OR 'cookie-file' or $HOME/.erlang.cookie will be read -->
<param name="cookie" value="ClueCon"/>
<!-- Read a cookie from an arbitary erlang cookie file instead -->
<!--<param name="cookie-file" value="/tmp/erlang.cookie"/>-->
<param name="shortname" value="true"/>
<!-- in additon to cookie, optionally restrict by ACL -->
<!--<param name="apply-inbound-acl" value="lan"/>-->
<!-- alternative is "binary" -->
<!--<param name="encoding" value="string"/>-->
<!-- provide compatability with previous OTP release (use with care) -->
<!--<param name="compat-rel" value="12"/> -->
</settings>
</configuration>
<configuration name="event_multicast.conf" description="Multicast Event">
<settings>
<param name="address" value="225.1.1.1"/>
<param name="port" value="4242"/>
<param name="bindings" value="all"/>
<param name="ttl" value="1"/>
<!-- <param name="loopback" value="no"/>-->
<!-- Uncomment this to enable pre-shared key encryption on the packets. -->
<!-- For this option to work, you'll need to have the openssl development -->
<!-- headers installed when you ran ./configure -->
<!-- <param name="psk" value="ClueCon"/> -->
</settings>
</configuration>
<configuration name="event_socket.conf" description="Socket Client">
<settings>
<param name="nat-map" value="false"/>
<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>
<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>
<configuration name="fifo.conf" description="FIFO Configuration">
<settings>
<param name="delete-all-outbound-member-on-startup" value="false"/>
</settings>
<fifos>
<fifo name="[email protected]" importance="0">
<!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/[email protected]</member>-->
</fifo>
</fifos>
</configuration>
<configuration name="hash.conf" description="Hash Configuration">
<remotes>
<!-- List of hosts from where to pull usage data -->
<!-- <remote name="Test1" host="10.0.0.10" port="8021" password="ClueCon" interval="1000" /> -->
</remotes>
</configuration>
<configuration name="httapi.conf" description="HT-TAPI Hypertext Telephony API">
<settings>
<!-- print xml on the consol -->
<param name="debug" value="true"/>
<!-- time to keep audio files when discoverd they were deleted from the http server -->
<param name="file-not-found-expires" value="300"/>
<!-- how often to re-check the server to make sure the remote file has not changed -->
<param name="file-cache-ttl" value="300"/>
</settings>
<profiles>
<profile name="default">
<!-- default params for conference action tags -->
<conference>
<param name="default-profile" value="default"/>
</conference>
<!-- default params for dial action tags -->
<dial>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
</dial>
<!-- permissions -->
<permissions>
<!-- <permission name="all" value="true"/> -->
<!--<permission name="none" value="true"/> -->
<permission name="set-params" value="true"/>
<permission name="set-vars" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
-->
</permission>
<permission name="get-vars" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
-->
</permission>
<permission name="extended-data" value="false"/>
<permission name="execute-apps" value="true">
<!-- default to "deny" or "allow" -->
<application-list default="deny">
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<application name="info"/>
<application name="hangup"/>
</application-list>
</permission>
<permission name="expand-vars-in-tag-body" value="false">
<!-- default to "deny" or "allow" -->
<!-- type attr can be "deny" or "allow" nothing defaults to opposite of the list default so allow in this case -->
<!--
<variable-list default="deny">