add freeswitch

Change-Id: If948de9e9e7e288b0a031ab973f453cd91a9e99b
This commit is contained in:
Oliver Smith 2019-11-08 18:16:30 +01:00 committed by Neels Hofmeyr
parent 87dae195e3
commit 667f19b199
82 changed files with 5244 additions and 10 deletions

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@ -110,6 +110,11 @@ SIPCON_LOCAL="${TO_SIP_IP}"
SIPCON_LOCAL_PORT="5060"
SIPCON_REMOTE="127.0.0.1"
SIPCON_REMOTE_PORT="5069"
# SIPCON_SERVER:
# "kamailio" -- launch kamailio
# "freeswitch" -- launch freeswitch
# "none" -- launch no server, use remotely running SIP server
SIPCON_SERVER="kamailio"
LOG_OUTPUT0_TYPE=stderr
LOG_OUTPUT1_TYPE=file current_log/${_name}.log

7
net/templates/freeswitch/.gitignore vendored Normal file
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@ -0,0 +1,7 @@
*
!*/
!.gitignore
!conf/**
!python/**
!/freeswitch.sh
__pycache__

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@ -0,0 +1,5 @@
<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>

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@ -0,0 +1,31 @@
<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"/>
</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="$${domain}"/>
<!-- use cidr= if you wish to allow ip ranges to this domains acl. -->
<!-- <node type="allow" cidr="192.168.0.0/24"/> -->
<node type="allow" cidr="127.0.0.2/32"/> <!-- FIXME: ${SIPCON_LOCAL} -->
</list>
</network-lists>
</configuration>

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@ -0,0 +1,12 @@
<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>

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@ -0,0 +1,18 @@
<configuration name="amr.conf" description="mod_amr configuration">
<settings name="default">
<!-- AMR mode set (see RFC 4867) -->
<!-- param name="mode-set" value="0,1,2,3,4,5,6,7" /-->
<param name="mode-set" value="4" />
<!-- enable octet alignment -->
<param name="octet-align" value="1"/>
<!-- rhizomatica custom param: force OA mode always for osmo-bts -->
<param name="force-oa" value="1"/>
<!-- enable discontinuous transmission -->
<param name="dtx" value="false"/>
<param name="default-bitrate" value="4"/>
</settings>
</configuration>

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@ -0,0 +1,11 @@
<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>

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@ -0,0 +1,38 @@
<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="$${hold_music}"/>
<!--<param name="record-template" value="$${base_dir}/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>

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@ -0,0 +1,13 @@
<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>

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@ -0,0 +1,18 @@
<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>

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<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>

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<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>

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@ -0,0 +1,213 @@
<!-- 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@$${domain}" status="FreeSWITCH"/>
</advertise>
<!-- These are the default keys that map when you do not specify a caller control group -->
<!-- Note: none and default are reserved names for group names. Disabled if dist-dtmf member flag is set. -->
<caller-controls>
<group name="default">
<control action="mute" digits="0"/>
<control action="deaf mute" digits="*"/>
<control action="energy up" digits="9"/>
<control action="energy equ" digits="8"/>
<control action="energy dn" digits="7"/>
<control action="vol talk up" digits="3"/>
<control action="vol talk zero" digits="2"/>
<control action="vol talk dn" digits="1"/>
<control action="vol listen up" digits="6"/>
<control action="vol listen zero" digits="5"/>
<control action="vol listen dn" digits="4"/>
<control action="hangup" digits="#"/>
</group>
</caller-controls>
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
<!--If no profile is specified it will default to "default"-->
<profile name="default">
<!-- Directory to drop CDR's
'auto' means $PREFIX/logs/conference_cdr/<confernece_uuid>.cdr.xml
a non-absolute path means $PREFIX/logs/<value>/<confernece_uuid>.cdr.xml
absolute path means <value>/<confernece_uuid>.cdr.xml
-->
<!-- <param name="cdr-log-dir" value="auto"/> -->
<!-- Domain (for presence) -->
<param name="domain" value="$${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|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="$${sounds_dir}/en/us/callie"/>-->
<!-- File to play to acknowledge succees -->
<!--<param name="ack-sound" value="beep.wav"/>-->
<!-- File to play to acknowledge failure -->
<!--<param name="nack-sound" value="beeperr.wav"/>-->
<!-- File to play to acknowledge muted -->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<!-- File to play to acknowledge unmuted -->
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<!-- File to play if you are alone in the conference -->
<param name="alone-sound" value="conference/conf-alone.wav"/>
<!-- File to play endlessly (nobody will ever be able to talk) -->
<!--<param name="perpetual-sound" value="perpetual.wav"/>-->
<!-- File to play when you're alone (music on hold)-->
<param name="moh-sound" value="$${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 are 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="$${outbound_caller_name}"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="$${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="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
<!-- IVR digit machine timeouts -->
<!-- How much to wait between DTMF digits to match caller-controls -->
<!-- <param name="ivr-dtmf-timeout" value="500"/> -->
<!-- How much to wait for the first DTMF, 0 forever -->
<!-- <param name="ivr-input-timeout" value="0" /> -->
<!-- Delay before a conference is asked to be terminated -->
<!-- <param name="endconf-grace-time" value="120" /> -->
<!-- Can be | delim of wait-mod|audio-always|video-bridge|video-floor-only
wait_mod will wait until the moderator in,
audio-always will always mix audio from all members regardless they are talking or not -->
<!-- <param name="conference-flags" value="audio-always"/> -->
</profile>
<profile name="wideband">
<param name="domain" value="$${domain}"/>
<param name="rate" value="16000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>-->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${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="$${domain}"/>
<param name="rate" value="32000"/>
<param name="interval" value="20"/>
<param name="energy-level" value="300"/>
<!--<param name="sound-prefix" value="$${sounds_dir}/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="$${sounds_dir}/en/us/callie"/>-->
<param name="muted-sound" value="conference/conf-muted.wav"/>
<param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
<param name="alone-sound" value="conference/conf-alone.wav"/>
<param name="moh-sound" value="$${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>
<profile name="sla">
<param name="domain" value="$${domain}"/>
<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>

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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>

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<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>

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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>

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<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
<param name="debug" value="0"/>
<param name="codec-prefs" value="H264,PCMU"/>
</settings>
<X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/>
</configuration>

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<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>

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<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>

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<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>

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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="auto-reload" value="true"/>
<param name="query-timeout-ms" value="200"/>
<param name="query-timeout-retry" value="2"/>
<param name="random-nameserver" value="false"/>
<!-- If you have specific (non-recursive) servers for your enum queries, specify them here ( up to 10 ) -->
<!-- <param name="nameserver" value="x.x.x.x"/> -->
<!-- <param name="nameserver" value="y.y.y.y"/> -->
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1;transport=udp"/>
<route service="E2T+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1;transport=tcp"/>
<!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
</routes>
</configuration>

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<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 freeswitch@example.com.
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>

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<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>

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<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>

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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>

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<configuration name="fifo.conf" description="FIFO Configuration">
<settings>
<param name="delete-all-outbound-member-on-startup" value="false"/>
</settings>
<fifos>
<fifo name="cool_fifo@$${domain}" importance="0">
<!--<member timeout="60" simo="1" lag="20">{member_wait=nowait}user/1005@$${domain}</member>-->
</fifo>
</fifos>
</configuration>

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<configuration name="format_cdr.conf" description="Multi Format CDR CURL logger">
<!-- You can have multiple profiles, to allow logging to both json and cdr simultaneously, or to
different paths or servers with different settings, just be sure to use different name for
each profile. -->
<profiles>
<profile name="default">
<settings>
<!-- the format of data to send, defaults to xml -->
<!-- <param name="format" value="json|xml"/> -->
<param name="format" value="xml"/>
<!-- 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"/> -->
<!-- Log via http and on disk, default is false -->
<!-- <param name="log-http-and-disk" value="true"/> -->
<!-- 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/format_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, 'base64' for base64 encoding or 'textxml' for text/xml -->
<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/format_cdr -->
<!-- <param name="err-log-dir" value="/tmp"/> -->
<!-- which auhtentification scheme to use. Supported values are: basic, digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${base_dir}/conf/certs/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${base_dir}/conf/certs/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${base_dir}/conf/certs/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="/tmp/cookie-mod_format_cdr_curl.txt"/> -->
<!-- Whether to URL encode the individual JSON values. Defaults to true, set to false for standard JSON. -->
<param name="encode-values" value="true"/>
</settings>
</profile>
</profiles>
</configuration>

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<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>

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<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">
<variable name="caller_id_name"/>
<variable name="hangup"/>
</variable-list>
<api-list default="deny">
<api name="expr"/>
<api name="lua"/>
</api-list>
-->
</permission>
<permission name="dial" value="true"/>
<permission name="dial-set-context" value="false"/>
<permission name="dial-set-dialplan" value="false"/>
<permission name="dial-set-cid-name" value="false"/>
<permission name="dial-set-cid-number" value="false"/>
<permission name="dial-full-originate" value="false"/>
<permission name="conference" value="true"/>
<permission name="conference-set-profile" value="false"/>
</permissions>
<params>
<!-- default url can be overridden by app data -->
<param name="gateway-url" value="http://www.freeswitch.org/api/index.cgi" />
<!-- set this to provide authentication credentials to the server -->
<!--<param name="gateway-credentials" value="muser:mypass"/>-->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${base_dir}/conf/certs/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${base_dir}/conf/certs/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional timeout -->
<!-- <param name="timeout" value="10"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${base_dir}/conf/certs/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/> -->
<!-- one or more of these imply you want to pick the exact variables that are transmitted -->
<!--<param name="enable-post-var" value="Caller-Unique-ID"/>-->
</params>
</profile>
</profiles>
</configuration>

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<configuration name="http_cache.conf" description="HTTP GET cache">
<settings>
<!-- set to true if you want to enable http:// and https:// formats. Do not use if mod_httapi is also loaded -->
<param name="enable-file-formats" value="false"/>
<param name="max-urls" value="10000"/>
<param name="location" value="$${base_dir}/http_cache"/>
<param name="default-max-age" value="86400"/>
<param name="prefetch-thread-count" value="8"/>
<param name="prefetch-queue-size" value="100"/>
<!-- absolute path to CA bundle file -->
<param name="ssl-cacert" value="$${base_dir}/conf/cacert.pem"/>
<!-- verify certificates -->
<param name="ssl-verifypeer" value="true"/>
<!-- verify host name matches certificate -->
<param name="ssl-verifyhost" value="true"/>
</settings>
</configuration>

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<configuration name="ivr.conf" description="IVR menus">
<menus>
<X-PRE-PROCESS cmd="include" data="../ivr_menus/*.xml"/>
</menus>
</configuration>

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<configuration name="java.conf" description="Java Plug-Ins">
<javavm path="/opt/jdk1.6.0_04/jre/lib/amd64/server/libjvm.so"/>
<options>
<option value="-Djava.class.path=$${base_dir}/scripts/freeswitch.jar:$${base_dir}/scripts/example.jar"/>
<option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=0.0.0.0:8000"/>
</options>
<startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/>
</configuration>

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<configuration name="lcr.conf" description="LCR Configuration">
<settings>
<param name="odbc-dsn" value="freeswitch-mysql:freeswitch:Fr33Sw1tch"/>
<!-- <param name="odbc-dsn" value="freeswitch-pgsql:freeswitch:Fr33Sw1tch"/> -->
</settings>
<profiles>
<profile name="default">
<param name="id" value="0"/>
<param name="order_by" value="rate,quality,reliability"/>
</profile>
<profile name="qual_rel">
<param name="id" value="1"/>
<param name="order_by" value="quality,reliability"/>
</profile>
<profile name="rel_qual">
<param name="id" value="2"/>
<param name="order_by" value="reliability,quality"/>
</profile>
<!--
Some samples of how to do custom SQL:
=============================================================
PostgreSQL with contrib prefix module which supports fast
prefix queries. Ideal option.
=============================================================
<profile name="pg_prefix">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> %q
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
=============================================================
PostgreSQL with contrib prefix module which supports fast
prefix queries. Ideal option. Alternate syntax which requies
a session but allows variable substitution.
=============================================================
<profile name="pg_prefix2">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits_prefix @> '${lcr_query_digits}'
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
=============================================================
Demonstrates use of computed inlist.
=============================================================
<profile name="inlist">
<param name="custom_sql" value="
SELECT l.digits AS lcr_digits,
c.carrier_name AS lcr_carrier_name,
l.${lcr_rate_field} as lcr_rate_field,
cg.prefix AS lcr_gw_prefix, cg.suffix AS lcr_gw_suffix,
l.lead_strip AS lcr_lead_strip, l.trail_strip AS lcr_trail_strip,
l.prefix AS lcr_prefix, l.suffix AS lcr_suffix
FROM lcr l
JOIN carriers c ON l.carrier_id=c.id
JOIN carrier_gateway cg ON c.id=cg.carrier_id
WHERE c.enabled = '1' AND cg.enabled = '1' AND l.enabled = '1'
AND digits IN (${lcr_query_expanded_digits})
AND CURRENT_TIMESTAMP BETWEEN date_start AND date_end
ORDER BY digits DESC, ${lcr_rate_field}, random();
"/>
</profile>
-->
</profiles>
</configuration>

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<configuration name="local_stream.conf" description="stream files from local dir">
<!-- fallback to default if requested moh class isn't found -->
<directory name="default" path="$${sounds_dir}/music/8000">
<param name="rate" value="8000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
<!-- list of short files to break in with every so often -->
<!--<param name="chime-list" value="file1.wav,file2.wav"/>-->
<!-- frequency of break-in (seconds)-->
<!--<param name="chime-freq" value="30"/>-->
<!-- limit to how many seconds the file will play -->
<!--<param name="chime-max" value="500"/>-->
</directory>
<directory name="moh/8000" path="$${sounds_dir}/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="$${sounds_dir}/music/16000">
<param name="rate" value="16000"/>
<param name="shuffle" value="true"/>
<param name="channels" value="1"/>
<param name="interval" value="20"/>
<param name="timer-name" value="soft"/>
</directory>
<directory name="moh/32000" path="$${sounds_dir}/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="$${sounds_dir}/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>

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<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"/>
<!-- Maximum number of log files to keep before wrapping -->
<!-- If this parameter is enabled, the log filenames will not include a date stamp -->
<!-- <param name="maximum-rotate" value="32"/> -->
<!-- Uncomment to prefix all log lines by the session's uuid -->
<!-- <param name="uuid" value="true" /> -->
</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>

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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"/>-->
<!--<hook event="CUSTOM" subclass="conference::maintenance" script="catch-event.lua"/>-->
</settings>
</configuration>

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<configuration name="memcache.conf" description="memcache Configuration">
<settings>
<!-- comma sep list of servers: eg: localhost,otherhost:port,anotherone -->
<param name="memcache-servers" value="localhost"/>
</settings>
</configuration>

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<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"/> -->
<!-- <load module="mod_xml_scgi"/> -->
<!-- Event Handlers -->
<!--<load module="mod_cdr_csv"/>-->
<load module="mod_cdr_pg_csv"/>
<!-- <load module="mod_cdr_sqlite"/> -->
<!-- <load module="mod_event_multicast"/> -->
<load module="mod_event_socket"/>
<!-- <load module="mod_event_zmq"/> -->
<!-- <load module="mod_zeroconf"/> -->
<!-- <load module="mod_erlang_event"/> -->
<!-- <load module="mod_snmp"/> -->
<!-- Directory Interfaces -->
<!-- <load module="mod_ldap"/> -->
<!-- Endpoints -->
<!-- <load module="mod_dingaling"/> -->
<!-- <load module="mod_portaudio"/> -->
<!-- <load module="mod_alsa"/> -->
<load module="mod_sofia"/>
<load module="mod_loopback"/>
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_freetdm"/> -->
<!-- <load module="mod_openzap"/> -->
<!-- <load module="mod_unicall"/> -->
<!-- <load module="mod_skinny"/> -->
<!-- <load module="mod_khomp"/> -->
<!-- <load module="mod_rtmp"/> -->
<load module="mod_sms"/>
<!-- Applications -->
<load module="mod_commands"/>
<!--<load module="mod_conference"/>-->
<!-- <load module="mod_curl"/> -->
<load module="mod_db"/>
<load module="mod_dptools"/>
<load module="mod_expr"/>
<load module="mod_fifo"/>
<load module="mod_hash"/>
<load module="mod_voicemail"/>
<!--<load module="mod_directory"/>-->
<!--<load module="mod_distributor"/>-->
<!--<load module="mod_lcr"/>-->
<load module="mod_esf"/>
<!--<load module="mod_fsv"/>-->
<!--<load module="mod_cluechoo"/>
<load module="mod_valet_parking"/>-->
<!--<load module="mod_fsk"/>-->
<!--<load module="mod_spy"/>-->
<!--<load module="mod_random"/>-->
<load module="mod_httapi"/>
<!--<load module="mod_translate"/>-->
<!-- SNOM Module -->
<!--<load module="mod_snom"/>-->
<!-- This one only works on Linux for now -->
<!--<load module="mod_ladspa"/>-->
<!-- Dialplan Interfaces -->
<!-- <load module="mod_dialplan_directory"/> -->
<load module="mod_dialplan_xml"/>
<!--<load module="mod_dialplan_asterisk"/>-->
<!-- Codec Interfaces -->
<load module="mod_spandsp"/>
<!--<load module="mod_g723_1"/>-->
<!--<load module="mod_bcg729"/>-->
<load module="mod_g729"/>
<load module="mod_amr"/>
<!--<load module="mod_ilbc"/>-->
<!--<load module="mod_speex"/>
<load module="mod_h26x"/>
<load module="mod_vp8"/>
<load module="mod_b64"/>-->
<!--<load module="mod_siren"/>-->
<!--<load module="mod_isac"/>-->
<!--<load module="mod_celt"/>-->
<!--<load module="mod_opus"/>-->
<!-- File Format Interfaces -->
<load module="mod_sndfile"/>
<load module="mod_native_file"/>
<!-- <load module="mod_shell_stream"/> -->
<!--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 -->
<!-- <load module="mod_timerfd"/> -->
<!-- <load module="mod_posix_timer"/> -->
<!-- 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_tts_commandline"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Say -->
<load module="mod_say_en"/>
<load module="mod_say_es"/>
<!-- <load module="mod_say_ru"/> -->
<!-- <load module="mod_say_zh"/> -->
<!-- Third party modules -->
<!--<load module="mod_nibblebill"/>-->
<!--<load module="mod_callcenter"/>-->
</modules>
</configuration>

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<configuration name="mongo.conf">
<settings>
<!--
connection-string handles different ways to connect to mongo
samples:
server:port
foo/server:port,server:port SET
-->
<param name="connection-string" value="127.0.0.1:27017"/>
<param name="min-connections" value="10"/>
<param name="max-connections" value="100"/>
<!--
<param name="map" value="function() { emit(this.a, 1); }"/>
<param name="reduce" value="function(key, values) { return Array.sum(values); }"/>
<param name="finalize" value="function(key, value) { return value;}"/>
-->
</settings>
</configuration>

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<configuration name="nibblebill.conf" description="Nibble Billing">
<settings>
<!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with these options -->
<!-- Information for connecting to your database -->
<param name="odbc-dsn" value="bandwidth.com"/>
<!-- The database table where your CASH column is located -->
<param name="db_table" value="accounts"/>
<!-- The column name where we store the value of the account -->
<param name="db_column_cash" value="cash"/>
<!-- The column name for the unique ID identifying the account -->
<param name="db_column_account" value="id"/>
<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
field nibble_balance is used for balance info
<param name="custom_sql_lookup" value="SELECT cash AS nibble_balance FROM accounts WHERE account_code='${nibble_account}'"/>
-->
<!-- Custom SQL for loading current balance - overrides column names
channel vars are interpreted.
nibble_increment is the amount to update
<param name="custom_sql_save" value="UPDATE accounts SET cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
-->
<!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
<param name="global_heartbeat" value="60"/>
<!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
<param name="lowbal_amt" value="5"/>
<param name="lowbal_action" value="play ding"/>
<!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
<param name="nobal_amt" value="0"/>
<param name="nobal_action" value="hangup"/>
<!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
<param name="percall_max_amt" value="100"/>
<param name="percall_action" value="hangup"/>
</settings>
</configuration>

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<configuration name="opal.conf" description="Opal Endpoints">
<settings>
<param name="trace-level" value="3"/>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-type" value="signal"/> <!-- string, signal, rfc2833, in-band -->
<param name="jitter-size" value="40,100"/> <!-- Jitter buffer min/max size, milliseconds -->
<!-- <param name="codec-prefs" value="PCMU,PCMA"/> --> <!-- list, and preferecnce order, of codecs -->
<!-- <param name="disable-transcoding" value="true"/> --> <!-- do not transcode, use source channel codec only -->
<param name="gk-address" value=""/> <!-- empty to disable, "*" to search LAN -->
<param name="gk-identifer" value=""/> <!-- optional name of gk -->
<param name="gk-interface" value="$${local_ip_v4}"/> <!-- optional listener interface name -->
</settings>
<listeners>
<listener name="default">
<param name="h323-ip" value="$${local_ip_v4}"/>
<param name="h323-port" value="1720"/>
</listener>
</listeners>
</configuration>

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<!--
To use this application simply install the open source Oreka recorder server (Orkaudio) and point
the sip-server-addr and sip-server-port to the oreka server
-->
<configuration name="oreka.conf" description="Oreka Recorder configuration">
<settings>
<!-- Oreka/Orkaudio recording server address -->
<!-- <param name="sip-server-addr" value="192.168.1.200"/> -->
<!-- Which port to send signaling to in the recording server -->
<!-- <param name="sip-server-port" value="6000"/> -->
</settings>
</configuration>

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<configuration name="osp.conf" description="OSP Module Configuration">
<settings>
<!-- Debug info flag -->
<param name="debug-info" value="disabled"/>
<!-- Log level for debug info -->
<param name="log-level" value="info"/>
<!-- Crypto hareware accelerate is disabled by default -->
<param name="crypto-hardware" value="disabled"/>
<!-- SIP settings -->
<param name="sip" module="sofia" profile="external"/>
<!-- H.323 settings -->
<!-- <param name="h323" module="h323" profile="external"/> -->
<!-- IAX settings -->
<!-- <param name="iax" module="iax" profile="external"/> -->
<!-- Skype settings -->
<!-- <param name="skype" module="skypopen" profile="external"/> -->
<!-- Default destination protocol -->
<param name="default-protocol" value="sip"/>
</settings>
<profiles>
<!-- Default OSP profile -->
<profile name="default">
<!-- Service point URLs, up to 8 allowed -->
<!-- <param name="service-point-url" value="http://osptestserver.transnexus.com:5045/osp"/> -->
<!-- <param name="service-point-url" value="https://127.0.0.1:1443/osp"/> -->
<param name="service-point-url" value="http://127.0.0.1:5045/osp"/>
<!-- FreeSWITCH IP address for OSP -->
<param name="device-ip" value="127.0.0.1:5080"/>
<!-- SSL lifetime in seconds -->
<param name="ssl-lifetime" value="300"/>
<!-- HTTP max connections, 1~1000 -->
<param name="http-max-connections" value="20"/>
<!-- HTTP persistence in seconds -->
<param name="http-persistence" value="60"/>
<!-- HTTP retry delay in seconds, 0~10 -->
<param name="http-retry-delay" value="0"/>
<!-- HTTP retry limit, 0~100 -->
<param name="http-retry-limit" value="2"/>
<!-- HTTP timeout in milliseconds, 200~60000 -->
<param name="http-timeout" value="10000"/>
<!-- OSP work mode, direct or indirect -->
<param name="work-mode" value="direct"/>
<!-- OSP service type, voice or npquery -->
<param name="service-type" value="voice"/>
<!-- Max number of destinations -->
<param name="max-destinations" value="12"/>
</profile>
</profiles>
</configuration>

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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>

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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="language-weight" value="1"/>-->
<!--<param name="narrowband-model" value="communicator"/>-->
<!--<param name="wideband-model" value="wsj1"/>-->
<!--<param name="dictionary" value="default.dic"/>-->
</settings>
</configuration>

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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"/>-->
<!--Enable or Disable dual_streams-->
<!--<param name="dual-streams" value="true"/>-->
<!--file to play when calls are on hold-->
<param name="hold-file" value="$${hold_music}"/>
<!--Timer to use for hold music (i'd leave this one commented)-->
<!--<param name="timer-name" value="soft"/>-->
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="$${outbound_caller_name}"/>
<param name="cid-num" value="$${outbound_caller_id}"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="48000"/>
<param name="codec-ms" value="20"/>
<!--uncomment the following line to make mod_portaudio fail to load if it fails to find a device-->
<!-- <param name="unload-on-device-fail" value="true"/> -->
</settings>
<!--
mod_portaudio "streams"
The portaudio streams were introduced to support multiple devices and multiple channels in mod_portaudio.
For example, if you have a sound card that supports multiple channels or have multiple sound cards and you
want to use them at the same time, you can do it configuring streams and endpoints here.
A "stream" is just a logical container for some settings required by portaudio in order to stream audio and
define a friendly name for that configuration. Streams in itself do not do anything else than contain configs.
Once you have your streams defined you can proceed to define "endpoints". Go to the "<endpoints>" section
for more information on endpoints.
You can use the command "pa shstreams" (portaudio shared streams) to show the configured streams.
-->
<streams>
<!--
In this example we define 2 streams, one for a usb audio device and another for the usual Mac defaults
The name="" attribute in the <stream> tag must uniquely identify the stream configuration and can be
later used when creating endpoints in the "instream" and "outstream" parameters of the endpoint.
-->
<!-- This sample "usb1" configuration was tested with a USB Griffin iMic device -->
<stream name="usb1">
<!--
Which device to use for input in this stream
The value for this parameter must be either in the form '#devno',
for example '#2' for device number 2, or 'device-name', like 'iMic USB audio system'
The output of command "pa devlist" will show you device names and numbers as enumerated
by portaudio.
-->
<param name="indev" value="#2" />
<!--
Same as the indev but for output. In this case the device is capable of input and output
Some devices are capable of input only or output only (see the default example)
-->
<param name="outdev" value="#2" />
<!-- The sample rate to use for this stream -->
<param name="sample-rate" value="48000" />
<!--
Size of the packets in milliseconds. The smaller the number the less latency you'll have
The minimum value is 10ms
-->
<param name="codec-ms" value="10" />
<!--
How many channels to open for this stream.
If you're device is stereo, you can choose 2 here. However, bear in mind that then
your left and right channels will be separated and when creating endpoints you will have
to either choose the left or right channel. This may or may not be what you want. This separation
means that you can have 2 separate FreeSWITCH calls, listening to one of them in your left channel
and the other in the right chanel.
-->
<param name="channels" value="2" />
</stream>
<!-- This default stream was tested using the default Macbook Pro input/output devices -->
<stream name="default">
<!-- The default system input device -->
<param name="indev" value="#0" />
<!-- The default system output device -->
<param name="outdev" value="#1" />
<!-- CD quality sampling rate ftw -->
<param name="sample-rate" value="48000" />
<!-- Low latency -->
<param name="codec-ms" value="10" />
<!-- Choosing 1 channel allows to hear in both left-right channel when using a headset -->
<param name="channels" value="1" />
</stream>
</streams>
<!--
mod_portaudio "endpoints"
Endpoints is a way to define the input and output that a given portaudio channel will use.
There is a lot of flexibility. You can create endpoints which are "send-only", which means
audio will be read from FreeSWITCH and sent down to the provided stream, but no audio will
be read from that stream and only silence provided back to FreeSWITCH.
send-only endpoint:
(FS CORE) ->-> audio ->-> sound-card-x
You can also create a read-only endpoint.
read-only-endpoint:
(FS CORE) <-<- audio <-<- sound-card-x
And of course you can create a bidirectional endpoint:
bidirectional-endpoint:
(FS CORE) <-> audio <-> sound-card-x
You can also define a stream which uses only the left or only the right channel of a given device stream.
This means you can have 2 SIP calls connected to the same device haring one call in your left ear and
the other call to your right ear :-)
The name="parameter" of the endpoint allows you to use it in the FreeSWITCH dial plan to dial, ie:
<action application="bridge" data="portaudio/endpoint/usb1out-left" />
You can use the command "pa endpoints" to show the configured endpoints.
-->
<endpoints>
<!--
An endpoint is a handle name to refer to a configuration that determines where to read media from
and write media to. The endpoint can use any input/output stream combination for that purpose as
long as the streams match the sampling rate and codec-ms (see <streams> XML tag).
You can also omit the instream or the outstream parameter (but obviously not both).
-->
<!--
Configuration for a "default" bidirectional endpoint that uses the default stream defined previously in
the <streams> section.
-->
<endpoint name="default">
<!--
The instream, outstream is the name of the stream and channel to use. The stream
name is the same you configured in the <streams> section. This parameters follow
the syntax <stream-name>:<channel index>. You can omit either the outstream
or the instream, but not both! The channel index is zero-based and must be consistent
with the number of channels available for that stream (as configured in the <stream> section).
You cannot use index 1 if you chose channels=1 in the stream configuration.
-->
<param name="instream" value="default:0" />
<param name="outstream" value="default:0" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'send-only' or 'output-only' and uses the channel index 0 (left channel in a stereo device)
-->
<endpoint name="usb1out-left">
<param name="outstream" value="usb1:0" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'send-only' or 'output-only' and uses the channel index 1 (right channel in a stereo device)
-->
<endpoint name="usb1out-right">
<param name="outstream" value="usb1:1" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'receive-only' or 'input-only' and uses the channel index 0 (left channel in a stereo device)
-->
<endpoint name="usb1in-left">
<param name="instream" value="usb1:0" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'receive-only' or 'input-only' and uses the channel index 1 (right channel in a stereo device)
-->
<endpoint name="usb1in-right">
<param name="instream" value="usb1:1" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'bidirectional' or 'send-receive' and uses the channel index 0 (left channel in a stereo device)
-->
<endpoint name="usb1-left">
<param name="instream" value="usb1:0" />
<param name="outstream" value="usb1:0" />
</endpoint>
<!--
This endpoint uses the USB stream defined previously in the <streams> section and
is 'bidirectional' or 'send-receive' and uses the channel index 1 (right channel in a stereo device)
-->
<endpoint name="usb1-right">
<param name="instream" value="usb1:1" />
<param name="outstream" value="usb1:1" />
</endpoint>
</endpoints>
</configuration>

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<configuration name="post_load_modules.conf" description="Modules">
<modules>
</modules>
</configuration>

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<configuration name="presence_map.conf" description="PRESENCE MAP">
<domains>
<domain name="$${domain}">
<exten regex="3\d+" proto="conf"/>
</domain>
</domains>
</configuration>

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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 module 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"/>-->
<!--<param name="startup-script" value="startup_script_2"/>-->
</settings>
</configuration>

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<configuration name="redis.conf" description="mod_redis Configuration">
<settings>
<param name="host" value="localhost"/>
<param name="port" value="6379"/>
<param name="timeout" value="10000"/>
</settings>
</configuration>

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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>

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<configuration name="rtmp.conf" description="RTMP Endpoint">
<profiles>
<profile name="default">
<settings>
<param name="bind-address" value="0.0.0.0:1935" />
<param name="context" value="public" />
<param name="dialplan" value="XML" />
<!-- If this is set to true, no unauthenticated inbound calls will be allowed -->
<param name="auth-calls" value="true" />
<!-- How much time should the clients buffer the media stream (miliseconds) -->
<param name="buffer-len" value="50" />
<!-- Sets the maximum size of outbound RTMP chunks -->
<param name="chunksize" value="512" />
</settings>
</profile>
</profiles>
</configuration>

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<configuration name="sangoma_codec.conf" description="Sangoma Codec Configuration">
<settings>
<!--
Comma separated list of codecs to register with FreeSWITCH,
by default (if this parameter is not set) all available codecs are registered.
Valid codec values are: PCMU,PCMA,G729,G726-32,G722,GSM,G723,AMR,G7221,iLBC
If this parameter is not specified only G729 will be registered
<param name="register" value="all"/>
-->
<!--
List of codecs to not register with FreeSWITCH, by default this is empty,
but you may want to not load PCMU and PCMA or may be others to not use your
resources in codecs that are done well and fast in software.
<param name="noregister" value="PCMU,PCMA"/>
-->
<!--
Transcoding SOAP server URL. If you are installing the soap server (sngtc_server)
in the same box where FreeSWITCH, do not use this value, the default URL
that is hard-coded will work out of the box for local installations.
If you modify this value, you must configure your SOAP server (/etc/sngtc/sngtc_server.conf.xml)
to listen for HTTP requests on the same IP/port that you specify here.
<param name="soapserver" value="http://192.168.1.100:8080"/>
-->
<!--
RTP IP to use
By default, this module asks FreeSWITCH for the local ip address. However if you want to use a specific
IP address you can set it here.
<param name="rtpip" value="192.168.1.1"/>
-->
</settings>
</configuration>

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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>

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<configuration name="skinny.conf" description="Skinny Endpoints">
<profiles>
<X-PRE-PROCESS cmd="include" data="../skinny_profiles/*.xml"/>
</profiles>
</configuration>

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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"/>
<!-- <param name="capture-server" value="udp:homer.domain.com:5060"/> -->
</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>

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<configuration name="spandsp.conf" description="SpanDSP config">
<modem-settings>
<!--
total-modems set to N will create that many soft-modems.
If you use them with Hylafax you need the following for each one numbered 0..N:
1) A line like this in /etc/inittab:
f0:2345:respawn:/usr/lib/fax/faxgetty /dev/FS0
2) copy conf/config.FS0 to /var/spool/hylafax/etc (or wherver the appropriate dir is on your system)
Subsequent modem configs would incrment the 0 to 1 and so on.
-->
<param name="total-modems" value="0"/>
<!-- Change the directory of the devices created from /dev. Needed if FS runs as non-root -->
<!-- <param name="directory" value="/dev/FS"/> -->
<!-- Default context and dialplan to use on inbound calls from the modems -->
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<!-- Extra tracing for debugging -->
<param name="verbose" value="false"/>
</modem-settings>
<fax-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"/>
</fax-settings>
<descriptors>
<!-- These tones are defined in Annex to ITU Operational Bulletin No. 781 - 1.II.2003 -->
<!-- Various Tones Used in National Networks (According to ITU-T Recommendation E.180)(03/1998) -->
<!-- North America -->
<descriptor name="1">
<tone name="CED_TONE">
<element freq1="2100" freq2="0" min="700" max="0"/>
</tone>
<tone name="SIT">
<element freq1="950" freq2="0" min="256" max="400"/>
<element freq1="1400" freq2="0" min="256" max="400"/>
<element freq1="1800" freq2="0" min="256" max="400"/>
</tone>
<tone name="RING_TONE" description="North America ring">
<element freq1="440" freq2="480" min="1200" max="0"/>
</tone>
<tone name="REORDER_TONE">
<element freq1="480" freq2="620" min="224" max="316"/>
<element freq1="0" freq2="0" min="168" max="352"/>
<element freq1="480" freq2="620" min="224" max="316"/>
</tone>
<tone name="BUSY_TONE">
<element freq1="480" freq2="620" min="464" max="536"/>
<element freq1="0" freq2="0" min="464" max="572"/>
<element freq1="480" freq2="620" min="464" max="536"/>
</tone>
</descriptor>
<!-- United Kingdom -->
<descriptor name="44">
<tone name="CED_TONE">
<element freq1="2100" freq2="0" min="500" max="0"/>
</tone>
<tone name="SIT">
<element freq1="950" freq2="0" min="256" max="400"/>
<element freq1="1400" freq2="0" min="256" max="400"/>
<element freq1="1800" freq2="0" min="256" max="400"/>
</tone>
<tone name="REORDER_TONE">
<element freq1="400" freq2="0" min="368" max="416"/>
<element freq1="0" freq2="0" min="336" max="368"/>
<element freq1="400" freq2="0" min="256" max="288"/>
<element freq1="0" freq2="0" min="512" max="544"/>
</tone>
<tone name="BUSY_TONE">
<element freq1="400" freq2="0" min="352" max="384"/>
<element freq1="0" freq2="0" min="352" max="384"/>
<element freq1="400" freq2="0" min="352" max="384"/>
<element freq1="0" freq2="0" min="352" max="384"/>
</tone>
</descriptor>
<!-- Germany -->
<descriptor name="49">
<tone name="CED_TONE">
<element freq1="2100" freq2="0" min="500" max="0"/>
</tone>
<tone name="SIT">
<element freq1="900" freq2="0" min="256" max="400"/>
<element freq1="1400" freq2="0" min="256" max="400"/>
<element freq1="1800" freq2="0" min="256" max="400"/>
</tone>
<tone name="REORDER_TONE">
<element freq1="425" freq2="0" min="224" max="272"/>
<element freq1="0" freq2="0" min="224" max="272"/>
</tone>
<tone name="BUSY_TONE">
<element freq1="425" freq2="0" min="464" max="516"/>
<element freq1="0" freq2="0" min="464" max="516"/>
</tone>
</descriptor>
</descriptors>
</configuration>

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<configuration name="switch.conf" description="Core Configuration">
<cli-keybindings>
<key name="1" value="help"/>
<key name="2" value="status"/>
<key name="3" value="show channels"/>
<key name="4" value="show calls"/>
<key name="5" value="sofia status"/>
<key name="6" value="reloadxml"/>
<key name="7" value="console loglevel 0"/>
<key name="8" value="console loglevel 7"/>
<key name="9" value="sofia status profile internal"/>
<key name="10" value="sofia profile internal siptrace on"/>
<key name="11" value="sofia profile internal siptrace off"/>
<key name="12" value="version"/>
</cli-keybindings>
<default-ptimes>
<!-- Set this to override the 20ms assumption of various codecs in the sdp with no ptime defined -->
<!-- <codec name="G729" ptime="40"/> -->
</default-ptimes>
<settings>
<!-- Colorize the Console -->
<param name="colorize-console" value="true"/>
<!-- Run the timer at 20ms by default and drop down as needed unless you set 1m-timer=true which was previous default -->
<!-- <param name="1ms-timer" value="true"/> -->
<!--
Set the Switch Name for HA environments.
When setting the switch name, it will override the system hostname for all DB and CURL requests
allowing cluster environments such as RHCS to have identical FreeSWITCH configurations but run
as different hostnames.
-->
<!-- <param name="switchname" value="freeswitch"/> -->
<!-- Maximum number of simultaneous DB handles open -->
<param name="max-db-handles" value="50"/>
<!-- Maximum number of seconds to wait for a new DB handle before failing -->
<param name="db-handle-timeout" value="10"/>
<!-- Minimum idle CPU before refusing calls -->
<!-- <param name="min-idle-cpu" value="25"/> -->
<!--
Max number of sessions to allow at any given time.
NOTICE: If you're driving 28 T1's in a single box you should set this to 644*2 or 1288
this will ensure you're able to use the entire DS3 without a problem. Otherwise you'll
be 144 channels short of always filling that DS3 up which can translate into waste.
-->
<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="info"/>
<!-- Set the core DEBUG level (0-10) -->
<!-- <param name="debug-level" value="10"/> -->
<!-- SQL Buffer length within rage of 32k to 10m -->
<!-- <param name="sql-buffer-len" value="1m"/> -->
<!-- Maximum SQL Buffer length must be greater than sql-buffer-len -->
<!-- <param name="max-sql-buffer-len" value="2m"/> -->
<!--
The min-dtmf-duration specifies the minimum DTMF duration to use on
outgoing events. Events shorter than this will be increased in duration
to match min_dtmf_duration. You cannot configure a dtmf duration on a
profile that is less than this setting. You may increase this value,
but cannot set it lower than 400. This value cannot exceed
max-dtmf-duration. -->
<!-- <param name="min-dtmf-duration" value="400"/> -->
<!--
The max-dtmf-duration caps the playout of a DTMF event at the specified
duration. Events exceeding this duration will be truncated to this
duration. You cannot configure a duration on a profile that exceeds
this setting. This setting can be lowered, but cannot exceed 192000.
This setting cannot be set lower than min_dtmf_duration. -->
<!-- <param name="max-dtmf-duration" value="192000"/> -->
<!--
The default_dtmf_duration specifies the DTMF duration to use on
originated DTMF events or on events that are received without a
duration specified. This value can be increased or lowered. This
value is lower-bounded by min_dtmf_duration and upper-bounded by
max-dtmf-duration\. -->
<!-- <param name="default-dtmf-duration" value="2000"/> -->
<!--
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"/>
Do not 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"/>
<!-- Enable verbose channel events to include every detail about a channel on every event -->
<!-- <param name="verbose-channel-events" value="no"/> -->
<!-- Enable clock nanosleep -->
<!-- <param name="enable-clock-nanosleep" value="true"/> -->
<!-- Enable monotonic timing -->
<!-- <param name="enable-monotonic-timing" value="true"/> -->
<!-- NEEDS DOCUMENTATION -->
<!-- <param name="enable-softtimer-timerfd" value="true"/> -->
<!-- <param name="enable-cond-yield" value="true"/> -->
<!-- <param name="enable-timer-matrix" value="true"/> -->
<!-- <param name="threaded-system-exec" value="true"/> -->
<!-- <param name="tipping-point" value="0"/> -->
<!-- <param name="timer-affinity" value="disabled"/> -->
<!-- NEEDS DOCUMENTATION -->
<!-- RTP port range -->
<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="16484"/>
<param name="rtp-enable-zrtp" value="false"/>
<!-- <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" /> -->
<!-- <param name="core-db-dsn" value="dsn:username:password" /> -->
<!-- The system will create all the db schemas automatically, set this to false to avoid this behaviour -->
<!-- <param name="auto-create-schemas" value="true"/> -->
<!-- <param name="auto-clear-sql" value="true"/> -->
<!-- <param name="enable-early-hangup" value="true"/> -->
<!-- <param name="core-dbtype" value="MSSQL"/> -->
<!-- Allow multiple registrations to the same account in the central registration table -->
<!-- <param name="multiple-registrations" value="true"/> -->
</settings>
</configuration>

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@ -0,0 +1,18 @@
<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="facility" value="user"/>
<param name="ident" value="freeswitch"/>
<param name="loglevel" value="warning"/>
<!-- Uncomment the following line to log uuids in syslogs (when applicable) -->
<!-- <param name="uuid" value="true"/> -->
</settings>
</configuration>

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<configuration name="timezones.conf" description="Timezones">
<timezones>
<zone name="Africa/Abidjan" value="GMT0" />
<zone name="Africa/Accra" value="GMT0" />
<zone name="Africa/Addis_Ababa" value="EAT-3" />
<zone name="Africa/Algiers" value="CET-1" />
<zone name="Africa/Asmara" value="EAT-3" />
<zone name="Africa/Asmera" value="EAT-3" />
<zone name="Africa/Bamako" value="GMT0" />
<zone name="Africa/Bangui" value="WAT-1" />
<zone name="Africa/Banjul" value="GMT0" />
<zone name="Africa/Bissau" value="GMT0" />
<zone name="Africa/Blantyre" value="CAT-2" />
<zone name="Africa/Brazzaville" value="WAT-1" />
<zone name="Africa/Bujumbura" value="CAT-2" />
<zone name="Africa/Cairo" value="EET-2EEST,M4.5.5/1,M9.5.5/3" />
<zone name="Africa/Casablanca" value="WET0" />
<zone name="Africa/Ceuta" value="CET-1CEST,M3.5.0,M10.5.0/3" />
<zone name="Africa/Conakry" value="GMT0" />
<zone name="Africa/Dakar" value="GMT0" />
<zone name="Africa/Dar_es_Salaam" value="EAT-3" />
<zone name="Africa/Djibouti" value="EAT-3" />
<zone name="Africa/Douala" value="WAT-1" />
<zone name="Africa/El_Aaiun" value="WET0" />
<zone name="Africa/Freetown" value="GMT0" />
<zone name="Africa/Gaborone" value="CAT-2" />
<zone name="Africa/Harare" value="CAT-2" />
<zone name="Africa/Johannesburg" value="SAST-2" />
<zone name="Africa/Kampala" value="EAT-3" />
<zone name="Africa/Khartoum" value="EAT-3" />
<zone name="Africa/Kigali" value="CAT-2" />
<zone name="Africa/Kinshasa" value="WAT-1" />
<zone name="Africa/Lagos" value="WAT-1" />
<zone name="Africa/Libreville" value="WAT-1" />
<zone name="Africa/Lome" value="GMT0" />
<zone name="Africa/Luanda" value="WAT-1" />
<zone name="Africa/Lubumbashi" value="CAT-2" />
<zone name="Africa/Lusaka" value="CAT-2" />
<zone name="Africa/Malabo" value="WAT-1" />
<zone name="Africa/Maputo" value="CAT-2" />
<zone name="Africa/Maseru" value="SAST-2" />
<zone name="Africa/Mbabane" value="SAST-2" />
<zone name="Africa/Mogadishu" value="EAT-3" />
<zone name="Africa/Monrovia" value="GMT0" />
<zone name="Africa/Nairobi" value="EAT-3" />
<zone name="Africa/Ndjamena" value="WAT-1" />
<zone name="Africa/Niamey" value="WAT-1" />
<zone name="Africa/Nouakchott" value="GMT0" />
<zone name="Africa/Ouagadougou" value="GMT0" />
<zone name="Africa/Porto-Novo" value="WAT-1" />
<zone name="Africa/Sao_Tome" value="GMT0" />
<zone name="Africa/Timbuktu" value="GMT0" />
<zone name="Africa/Tripoli" value="EET-2" />
<zone name="Africa/Tunis" value="CET-1CEST,M3.5.0,M10.5.0/3" />
<zone name="Africa/Windhoek" value="WAT-1WAST,M9.1.0,M4.1.0" />
<zone name="America/Adak" value="HAST10HADT,M3.2.0,M11.1.0" />
<zone name="America/Anchorage" value="AKST9AKDT,M3.2.0,M11.1.0" />
<zone name="America/Anguilla" value="AST4" />
<zone name="America/Antigua" value="AST4" />
<zone name="America/Araguaina" value="BRT3" />
<zone name="America/Argentina/Buenos_Aires" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Catamarca" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/ComodRivadavia" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Cordoba" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Jujuy" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/La_Rioja" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Mendoza" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Rio_Gallegos" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/San_Juan" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/San_Luis" value="ART3" />
<zone name="America/Argentina/Tucuman" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Argentina/Ushuaia" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Aruba" value="AST4" />
<zone name="America/Asuncion" value="PYT4PYST,M10.3.0/0,M3.2.0/0" />
<zone name="America/Atikokan" value="EST5" />
<zone name="America/Atka" value="HAST10HADT,M3.2.0,M11.1.0" />
<zone name="America/Bahia" value="BRT3" />
<zone name="America/Barbados" value="AST4" />
<zone name="America/Belem" value="BRT3" />
<zone name="America/Belize" value="CST6" />
<zone name="America/Blanc-Sablon" value="AST4" />
<zone name="America/Boa_Vista" value="AMT4" />
<zone name="America/Bogota" value="COT5" />
<zone name="America/Boise" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/Buenos_Aires" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Cambridge_Bay" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/Campo_Grande" value="AMT4AMST,M10.2.0/0,M2.3.0/0" />
<zone name="America/Cancun" value="CST6CDT,M4.1.0,M10.5.0" />
<zone name="America/Caracas" value="VET4:30" />
<zone name="America/Catamarca" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Cayenne" value="GFT3" />
<zone name="America/Cayman" value="EST5" />
<zone name="America/Chicago" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Chihuahua" value="MST7MDT,M4.1.0,M10.5.0" />
<zone name="America/Coral_Harbour" value="EST5" />
<zone name="America/Cordoba" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Costa_Rica" value="CST6" />
<zone name="America/Cuiaba" value="AMT4AMST,M10.2.0/0,M2.3.0/0" />
<zone name="America/Curacao" value="AST4" />
<zone name="America/Danmarkshavn" value="GMT0" />
<zone name="America/Dawson" value="PST8PDT,M3.2.0,M11.1.0" />
<zone name="America/Dawson_Creek" value="MST7" />
<zone name="America/Denver" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/Detroit" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Dominica" value="AST4" />
<zone name="America/Edmonton" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/Eirunepe" value="ACT5" />
<zone name="America/El_Salvador" value="CST6" />
<zone name="America/Ensenada" value="PST8PDT,M4.1.0,M10.5.0" />
<zone name="America/Fortaleza" value="BRT3" />
<zone name="America/Fort_Wayne" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Glace_Bay" value="AST4ADT,M3.2.0,M11.1.0" />
<zone name="America/Godthab" value="WGST" />
<zone name="America/Goose_Bay" value="AST4ADT,M3.2.0/0:01,M11.1.0/0:01" />
<zone name="America/Grand_Turk" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Grenada" value="AST4" />
<zone name="America/Guadeloupe" value="AST4" />
<zone name="America/Guatemala" value="CST6" />
<zone name="America/Guayaquil" value="ECT5" />
<zone name="America/Guyana" value="GYT4" />
<zone name="America/Halifax" value="AST4ADT,M3.2.0,M11.1.0" />
<zone name="America/Havana" value="CST5CDT,M3.3.0/0,M10.5.0/1" />
<zone name="America/Hermosillo" value="MST7" />
<zone name="America/Indiana/Indianapolis" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Knox" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Marengo" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Petersburg" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indianapolis" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Tell_City" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Vevay" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Vincennes" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Indiana/Winamac" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Inuvik" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/Iqaluit" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Jamaica" value="EST5" />
<zone name="America/Jujuy" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Juneau" value="AKST9AKDT,M3.2.0,M11.1.0" />
<zone name="America/Kentucky/Louisville" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Kentucky/Monticello" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Knox_IN" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/La_Paz" value="BOT4" />
<zone name="America/Lima" value="PET5" />
<zone name="America/Los_Angeles" value="PST8PDT,M3.2.0,M11.1.0" />
<zone name="America/Louisville" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Maceio" value="BRT3" />
<zone name="America/Managua" value="CST6" />
<zone name="America/Manaus" value="AMT4" />
<zone name="America/Marigot" value="AST4" />
<zone name="America/Martinique" value="AST4" />
<zone name="America/Mazatlan" value="MST7MDT,M4.1.0,M10.5.0" />
<zone name="America/Mendoza" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Menominee" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Merida" value="CST6CDT,M4.1.0,M10.5.0" />
<zone name="America/Mexico_City" value="CST6CDT,M4.1.0,M10.5.0" />
<zone name="America/Miquelon" value="PMST3PMDT,M3.2.0,M11.1.0" />
<zone name="America/Moncton" value="AST4ADT,M3.2.0,M11.1.0" />
<zone name="America/Monterrey" value="CST6CDT,M4.1.0,M10.5.0" />
<zone name="America/Montevideo" value="UYT3UYST,M10.1.0,M3.2.0" />
<zone name="America/Montreal" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Montserrat" value="AST4" />
<zone name="America/Nassau" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/New_York" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Nipigon" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Nome" value="AKST9AKDT,M3.2.0,M11.1.0" />
<zone name="America/Noronha" value="FNT2" />
<zone name="America/North_Dakota/Center" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/North_Dakota/New_Salem" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Panama" value="EST5" />
<zone name="America/Pangnirtung" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="America/Paramaribo" value="SRT3" />
<zone name="America/Phoenix" value="MST7" />
<zone name="America/Port-au-Prince" value="EST5" />
<zone name="America/Porto_Acre" value="ACT5" />
<zone name="America/Port_of_Spain" value="AST4" />
<zone name="America/Porto_Velho" value="AMT4" />
<zone name="America/Puerto_Rico" value="AST4" />
<zone name="America/Rainy_River" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Rankin_Inlet" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="America/Recife" value="BRT3" />
<zone name="America/Regina" value="CST6" />
<zone name="America/Resolute" value="EST5" />
<zone name="America/Rio_Branco" value="ACT5" />
<zone name="America/Rosario" value="ART3ARST,M10.1.0/0,M3.3.0/0" />
<zone name="America/Santiago" value="CLST" />
<zone name="America/Santo_Domingo" value="AST4" />
<zone name="America/Sao_Paulo" value="BRT3BRST,M10.2.0/0,M2.3.0/0" />
<zone name="America/Scoresbysund" value="EGT1EGST,M3.5.0/0,M10.5.0/1" />
<zone name="America/Shiprock" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="America/St_Barthelemy" value="AST4" />
<zone name="America/St_Johns" value="NST3:30NDT,M3.2.0/0:01,M11.1.0/0:01" />
<zone name="America/St_Kitts" value="AST4" />
<zone name="America/St_Lucia" value="AST4" />
<zone name="America/St_Thomas" value="AST4" />
<zone name="America/St_Vincent" value="AST4" />
<zone name="America/Swift_Current" value="CST6" />
<zone name="America/Tegucigalpa" value="CST6" />
<zone name="America/Thule" value="AST4ADT,M3.2.0,M11.1.0" />
<zone name="America/Thunder_Bay" value="EST5EDT,M3.2.0,M11.1.0" />
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<zone name="Pacific/Nauru" value="NRT-12" />
<zone name="Pacific/Niue" value="NUT11" />
<zone name="Pacific/Norfolk" value="NFT-11:30" />
<zone name="Pacific/Noumea" value="NCT-11" />
<zone name="Pacific/Pago_Pago" value="SST11" />
<zone name="Pacific/Palau" value="PWT-9" />
<zone name="Pacific/Pitcairn" value="PST8" />
<zone name="Pacific/Ponape" value="PONT-11" />
<zone name="Pacific/Port_Moresby" value="PGT-10" />
<zone name="Pacific/Rarotonga" value="CKT10" />
<zone name="Pacific/Saipan" value="ChST-10" />
<zone name="Pacific/Samoa" value="SST11" />
<zone name="Pacific/Tahiti" value="TAHT10" />
<zone name="Pacific/Tarawa" value="GILT-12" />
<zone name="Pacific/Tongatapu" value="TOT-13" />
<zone name="Pacific/Truk" value="TRUT-10" />
<zone name="Pacific/Wake" value="WAKT-12" />
<zone name="Pacific/Wallis" value="WFT-12" />
<zone name="Pacific/Yap" value="TRUT-10" />
<zone name="SystemV/AST4" value="AST4" />
<zone name="SystemV/AST4ADT" value="AST4ADT,M3.2.0,M11.1.0" />
<zone name="SystemV/CST6" value="CST6" />
<zone name="SystemV/CST6CDT" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="SystemV/EST5" value="EST5" />
<zone name="SystemV/EST5EDT" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="SystemV/HST10" value="HST10" />
<zone name="SystemV/MST7" value="MST7" />
<zone name="SystemV/MST7MDT" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="SystemV/PST8" value="PST8" />
<zone name="SystemV/PST8PDT" value="PST8PDT,M3.2.0,M11.1.0" />
<zone name="SystemV/YST9" value="GAMT9" />
<zone name="SystemV/YST9YDT" value="AKST9AKDT,M3.2.0,M11.1.0" />
<zone name="US/Alaska" value="AKST9AKDT,M3.2.0,M11.1.0" />
<zone name="US/Aleutian" value="HAST10HADT,M3.2.0,M11.1.0" />
<zone name="US/Arizona" value="MST7" />
<zone name="US/Central" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="US/Eastern" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="US/East-Indiana" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="US/Hawaii" value="HST10" />
<zone name="US/Indiana-Starke" value="CST6CDT,M3.2.0,M11.1.0" />
<zone name="US/Michigan" value="EST5EDT,M3.2.0,M11.1.0" />
<zone name="US/Mountain" value="MST7MDT,M3.2.0,M11.1.0" />
<zone name="US/Pacific" value="PST8PDT,M3.2.0,M11.1.0" />
<zone name="US/Samoa" value="SST11" />
</timezones>
</configuration>

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<include>
<configuration name="translate.conf" description="Number Translation Rules">
<profiles>
<profile name="US">
<rule regex="^\+(\d+)$" replace="$1"/>
<rule regex="^(1[2-9]\d{2}[2-9]\d{6})$" replace="$1"/>
<rule regex="^([2-9]\d{2}[2-9]\d{6})$" replace="1$1"/>
<rule regex="^([2-9]\d{6})$" replace="1${areacode}$1"/>
<rule regex="^011(\d+)$" replace="$1"/>
</profile>
<profile name="GB">
<rule regex="^\+(\d+)$" replace="$1"/>
<rule regex="^$" replace="$1"/>
</profile>
<profile name="HK">
<rule regex="\+(\d+)$" replace="$1"/>
<rule regex="^(852\d{8})$" replace="$1"/>
<rule regex="^(\d{8})$" replace="852$1"/>
</profile>
</profiles>
<!--
<profile name="">
<rule regex="^\+(\d+)$" replace="$1"/>
<rule regex="^$" replace="$1"/>
</profile>
-->
</configuration>
</include>

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<configuration name="tts_commandline.conf" description="TextToSpeech Commandline configuration">
<settings>
<!--
Some variables will be replaced :
${text}: input text (quoted)
${rate}: sample rate (example: 8000)
${voice}: voice_name passed to TTS(quoted)
${file}: output file (quoted, including .wav extension)
Example commands can be found at:
http://wiki.freeswitch.org/wiki/Mod_tts_commandline#Example_commands
-->
<param name="command" value="echo ${text} | text2wave -f ${rate} > ${file}"/>
</settings>
</configuration>

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<configuration name="unicall.conf" description="Unicall Configuration">
<settings>
<param name="context" value="default"/>
<param name="dialplan" value="XML"/>
<param name="suppress-dtmf-tone" value="true"/>
</settings>
<spans>
<span id="1">
<param name="protocol-class" value="mfcr2"/>
<param name="protocol-variant" value="ar"/>
<param name="protocol-end" value="peer"/>
<param name="outgoing-allowed" value="true"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
<span id="2">
<param name="protocol-class" value="mfcr2"/>
<param name="protocol-variant" value="ar"/>
<param name="protocol-end" value="peer"/>
<param name="outgoing-allowed" value="true"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
</span>
</spans>
</configuration>

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<configuration name="unimrcp.conf" description="UniMRCP Client">
<settings>
<!-- UniMRCP profile to use for TTS -->
<param name="default-tts-profile" value="voxeo-prophecy8.0-mrcp1"/>
<!-- UniMRCP profile to use for ASR -->
<param name="default-asr-profile" value="voxeo-prophecy8.0-mrcp1"/>
<!-- UniMRCP logging level to appear in freeswitch.log. Options are:
EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG -->
<param name="log-level" value="DEBUG"/>
<!-- Enable events for profile creation, open, and close -->
<param name="enable-profile-events" value="false"/>
<param name="max-connection-count" value="100"/>
<param name="offer-new-connection" value="1"/>
<param name="request-timeout" value="3000"/>
</settings>
<profiles>
<X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/>
</profiles>
</configuration>

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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="max-retries" value="3"/>
<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"/>
<!-- play-new-messages-lifo and play-saved-messages-lifo default is false, playing oldest messages first
<param name="play-new-messages-lifo" value="false"/>
<param name="play-saved-messages-lifo" value="false"/>
-->
<param name="login-keys" value="0"/>
<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="skip-greet-key" value="#"/>
<param name="previous-message-key" value="1"/>
<param name="next-message-key" value="3"/>
<param name="skip-info-key" value="*"/>
<param name="repeat-message-key" value="0"/>
<param name="record-silence-threshold" value="200"/>
<param name="record-silence-hits" value="2"/>
<param name="web-template-file" value="web-vm.tpl"/>
<param name="db-password-override" value="false"/>
<param name="allow-empty-password-auth" value="true"/>
<!-- 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>

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<configuration name="voicemail_ivr.conf" description="Voicemail IVR">
<profiles>
<profile name="default">
<settings>
<param name="IVR-Maximum-Attempts" value="3" />
<param name="IVR-Entry-Timeout" value="3000" />
<param name="Record-Format" value="wav" />
<!--<param name="Record-Sample-Rate" value="8000" />-->
<param name="Record-Silence-Hits" value="4" />
<param name="Record-Silence-Threshold" value="200" />
<param name="Record-Maximum-Length" value="30" />
<!--<param name="Record-Minimum-Length" value="3" />-->
<param name="Exit-Purge" value="true" />
<param name="Password-Mask" value="XXX." />
<param name="User-Mask" value="X." />
</settings>
<apis>
<api name="auth_login" value="vm_fsdb_auth_login" />
<api name="msg_list" value="vm_fsdb_msg_list" />
<api name="msg_count" value="vm_fsdb_msg_count" />
<api name="msg_delete" value="vm_fsdb_msg_delete" />
<api name="msg_undelete" value="vm_fsdb_msg_undelete" />
<api name="msg_save" value="vm_fsdb_msg_save" />
<api name="msg_purge" value="vm_fsdb_msg_purge" />
<api name="msg_get" value="vm_fsdb_msg_get" />
<api name="msg_forward" value="vm_fsdb_msg_forward" />
<api name="pref_greeting_set" value="vm_fsdb_pref_greeting_set" />
<api name="pref_greeting_get" value="vm_fsdb_pref_greeting_get" />
<api name="pref_recname_set" value="vm_fsdb_pref_recname_set" />
<api name="pref_password_set" value="vm_fsdb_pref_password_set" />
</apis>
<menus>
<menu name="std_authenticate">
<phrases>
<phrase name="fail_auth" value="fail_auth@voicemail_ivr" />
</phrases>
<keys>
</keys>
</menu>
<menu name="std_authenticate_ask_user">
<phrases>
<phrase name="instructions" value="enter_id@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
</keys>
</menu>
<menu name="std_authenticate_ask_password">
<phrases>
<phrase name="instructions" value="enter_pass@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
</keys>
</menu>
<menu name="std_main_menu">
<settings>
<param name="Action-On-New-Message" value="new_msg:std_navigator" />
</settings>
<phrases>
<phrase name="msg_count" value="message_count@voicemail_ivr" />
<phrase name="say_date" value="say_date_event@voicemail_ivr" />
<phrase name="say_msg_number" value="say_message_number@voicemail_ivr" />
<phrase name="menu_options" value="menu@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="new_msg:std_navigator" variable="VM-Key-Play-New-Messages" />
<key dtmf="2" action="saved_msg:std_navigator" variable="VM-Key-Play-Saved-Messages" />
<key dtmf="5" action="menu:std_preference" variable="VM-Key-Config-Menu"/>
<key dtmf="#" action="return" variable="VM-Key-Terminator" />
</keys>
</menu>
<menu name="std_navigator">
<settings>
<!--<param name="Nav-Action-On-Delete" value="next_msg" />-->
</settings>
<phrases>
<phrase name="msg_count" value="message_count@voicemail_ivr" />
<phrase name="say_date" value="say_date_event@voicemail_ivr" />
<phrase name="say_msg_number" value="say_message_number@voicemail_ivr" />
<phrase name="menu_options" value="listen_file_check@voicemail_ivr" />
<phrase name="ack" value="ack@voicemail_ivr" />
<phrase name="play_message" value="play_message@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="skip_intro" variable="VM-Key-Main-Listen-File" />
<key dtmf="6" action="next_msg" variable="VM-Key-Main-Next-Msg" />
<key dtmf="4" action="prev_msg" />
<key dtmf="7" action="delete_msg" variable="VM-Key-Main-Delete-File" /> <!-- Same key for undelete if it already deleted -->
<key dtmf="8" action="menu:std_forward" variable="VM-Key-Main-Forward" />
<key dtmf="2" action="save_msg" variable="VM-Key-Main-Save-File" />
<key dtmf="5" action="callback" variable="VM-Key-Main-Callback" />
<key dtmf="#" action="return" /> <!-- TODO Might Conflict with future fast-forward -->
</keys>
</menu>
<menu name="std_preference">
<phrases>
<phrase name="menu_options" value="config_menu@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="menu:std_record_greeting_with_slot" variable="VM-Key-Record-Greeting" />
<key dtmf="2" action="menu:std_select_greeting_slot" variable="VM-Key-Choose-Greeting" />
<key dtmf="3" action="menu:std_record_name" variable="VM-Key-Record-Name" />
<key dtmf="6" action="menu:std_set_password" variable="VM-Key-Change-Password" />
<key dtmf="0" action="return" variable="VM-Key-Main-Menu" />
</keys>
</menu>
<menu name="std_record_greeting">
<phrases>
<phrase name="instructions" value="record_greeting@voicemail_ivr" />
<phrase name="play_recording" value="play_recording@voicemail_ivr" />
<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
<key dtmf="2" action="save" variable="VM-Key-Save-File" />
<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
<key dtmf="#" action="skip_instruction" />
</keys>
</menu>
<menu name="std_record_name">
<phrases>
<phrase name="instructions" value="record_name@voicemail_ivr" />
<phrase name="play_recording" value="play_recording@voicemail_ivr" />
<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
<key dtmf="2" action="save" variable="VM-Key-Save-File" />
<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
<key dtmf="#" action="skip_instruction" />
</keys>
</menu>
<menu name="std_record_message">
<phrases>
<phrase name="instructions" value="record_message@voicemail_ivr" />
<phrase name="play_recording" value="play_recording@voicemail_ivr" />
<phrase name="menu_options" value="record_file_check@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="listen" variable="VM-Key-Listen-File" />
<key dtmf="2" action="save" variable="VM-Key-Save-File" />
<key dtmf="4" action="rerecord" variable="VM-Key-ReRecord-File" />
<key dtmf="#" action="skip_instruction" />
</keys>
</menu>
<menu name="std_forward_ask_prepend">
<phrases>
<phrase name="menu_options" value="forward_ask_prepend@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="1" action="prepend" variable="VM-Key-Prepend" />
<key dtmf="8" action="forward" variable="VM-Key-Forward" />
<key dtmf="#" action="return" variable="VM-Key-Return" />
</keys>
</menu>
<menu name="std_forward_ask_extension">
<phrases>
<phrase name="instructions" value="forward_ask_extension@voicemail_ivr" />
<phrase name="ack" value="ack@voicemail_ivr" />
<phrase name="invalid_extension" value="invalid_extension@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
</keys>
</menu>
<menu name="std_select_greeting_slot">
<phrases>
<phrase name="instructions" value="choose_greeting@voicemail_ivr" />
<phrase name="invalid_slot" value="choose_greeting_fail@voicemail_ivr" />
<phrase name="selected_slot" value="greeting_selected@voicemail_ivr" />
</phrases>
<keys>
</keys>
</menu>
<menu name="std_record_greeting_with_slot">
<phrases>
<phrase name="instructions" value="choose_greeting@voicemail_ivr" />
</phrases>
<keys>
</keys>
</menu>
<menu name="std_set_password">
<phrases>
<phrase name="instructions" value="enter_pass@voicemail_ivr" />
</phrases>
<keys>
<key dtmf="#" action="ivrengine:terminate_entry" variable="VM-Key-Terminator" />
</keys>
</menu>
</menus>
</profile>
</profiles>
</configuration>

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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"/> -->
<!-- Log via http and on disk, default is false -->
<!-- <param name="log-http-and-disk" value="true"/> -->
<!-- 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, 'base64' for base64 encoding or 'textxml' for text/xml -->
<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"/> -->
<!-- which auhtentification scheme to use. Supported values are: basic, digest, NTLM, GSS-NEGOTIATE or "any" for automatic detection -->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${base_dir}/conf/certs/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${base_dir}/conf/certs/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${base_dir}/conf/certs/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/> -->
</settings>
</configuration>

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<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings>
<binding name="example">
<!-- Allow to bind on a particular IP for requests sent -->
<!--<param name="bind-local" value="$${local_ip_v4}" />-->
<!-- The url to a gateway cgi that can generate xml similar to
what's in this file only on-the-fly (leave it commented if you dont
need it) -->
<!-- one or more |-delim of configuration|directory|dialplan -->
<!-- <param name="gateway-url" value="http://www.freeswitch.org/gateway.xml" bindings="dialplan"/> -->
<!-- set this to provide authentication credentials to the server -->
<!--<param name="gateway-credentials" value="muser:mypass"/>-->
<!--<param name="auth-scheme" value="basic"/>-->
<!-- optional: this will enable the CA root certificate check by libcurl to
verify that the certificate was issued by a major Certificate Authority.
note: default value is disabled. only enable if you want this! -->
<!--<param name="enable-cacert-check" value="true"/>-->
<!-- optional: verify that the server is actually the one listed in the cert -->
<!-- <param name="enable-ssl-verifyhost" value="true"/> -->
<!-- optional: these options can be used to specify custom SSL certificates
to use for HTTPS communications. Either use both options or neither.
Specify your public key with 'ssl-cert-path' and the private key with
'ssl-key-path'. If your private key has a password, specify it with
'ssl-key-password'. -->
<!-- <param name="ssl-cert-path" value="$${base_dir}/conf/certs/public_key.pem"/> -->
<!-- <param name="ssl-key-path" value="$${base_dir}/conf/certs/private_key.pem"/> -->
<!-- <param name="ssl-key-password" value="MyPrivateKeyPassword"/> -->
<!-- optional timeout -->
<!-- <param name="timeout" value="10"/> -->
<!-- optional: use a custom CA certificate in PEM format to verify the peer
with. This is useful if you are acting as your own certificate authority.
note: only makes sense if used in combination with "enable-cacert-check." -->
<!-- <param name="ssl-cacert-file" value="$${base_dir}/conf/certs/cacert.pem"/> -->
<!-- optional: specify the SSL version to force HTTPS to use. Valid options are
"SSLv3" and "TLSv1". Otherwise libcurl will auto-negotiate the version. -->
<!-- <param name="ssl-version" value="TLSv1"/> -->
<!-- optional: enables cookies and stores them in the specified file. -->
<!-- <param name="cookie-file" value="/tmp/cookie-mod_xml_curl.txt"/> -->
<!-- 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>

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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>

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<configuration name="xml_scgi.conf" description="SCGI XML Gateway">
<bindings>
<binding name="example">
<!-- one or more |-delim of configuration|directory|dialplan -->
<!-- <param name="host" value="127.0.0.1" bindings="dialplan"/> -->
<!-- <param name="port" value="8080"/> -->
<!-- <param name="timeout" value="10"/> -->
<!-- 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>

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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>

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<?xml version="1.0" encoding="utf-8"?>
<include>
<context name="public">
<extension name="dialplan-dgsm">
<condition field="to" expression="^(.*)$">
<action application="python" data="dialplan-dgsm"/>
</condition>
</extension>
</context>
</include>

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<!--
NOTICE:
This context is usually accessed via the external sip profile listening 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="${destination_number}"/>
</condition>
</extension>
<!--
Tag anything pass thru here as an outside_call so you can make sure not
to create any routing loops based on the conditions that it came from
the outside of the switch.
-->
<extension name="outside_call" continue="true">
<condition>
<action application="set" data="outside_call=true"/>
<action application="export" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
</condition>
</extension>
<extension name="call_debug" continue="true">
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</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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[default]
; Things you're used to....
exten => music,n,Dial(SIP/1234@conference.freeswitch.org|120)
exten => _1XXXXX,n,set(cool=${EXTEN})
exten => _1XXXXX,n,set(myvar=true)
exten => _1XXXXX,n,Goto(default|music)
exten => 2137991400/1000,n,Goto(default|music)
; Some new magic you can do....
exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,enum($1)
exten => ~^(18(0{2}|8{2}|7{2}|6{2})\d{7})$,n,bridge(${enum_auto_route})
; instead of exten, put anything about the call you would rather match on.
; either the names of a field in caller_profile or a string of variables to expand.
caller_id_number => 2137991400,n,Goto(default|music)
${sip_from_user} => bill,n,Goto(default|music)

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<?xml version="1.0"?>
<!--
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">
<!--#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.
-->
<!--#comment
vars.xml contains all the #set directives for the preprocessor.
-->
<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>
<section name="chatplan" description="Regex/XML Chatplan">
<X-PRE-PROCESS cmd="include" data="chatplan/*.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>
</document>

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# This is a comment. I love comments.
# This file controls what Internet media types are sent to the client for
# given file extension(s). Sending the correct media type to the client
# is important so they know how to handle the content of the file.
# Extra types can either be added here or by using an AddType directive
# in your config files. For more information about Internet media types,
# please read RFC 2045, 2046, 2047, 2048, and 2077. The Internet media type
# registry is at <http://www.iana.org/assignments/media-types/>.
# MIME type Extensions
application/activemessage
application/andrew-inset ez
application/applefile
application/atom+xml atom
application/atomcat+xml atomcat
application/atomicmail
application/atomsvc+xml atomsvc
application/auth-policy+xml
application/batch-smtp
application/beep+xml
application/cals-1840
application/ccxml+xml ccxml
application/cellml+xml
application/cnrp+xml
application/commonground
application/conference-info+xml
application/cpl+xml
application/csta+xml
application/cstadata+xml
application/cybercash
application/davmount+xml davmount
application/dca-rft
application/dec-dx
application/dialog-info+xml
application/dicom
application/dns
application/dvcs
application/ecmascript ecma
application/edi-consent
application/edi-x12
application/edifact
application/epp+xml
application/eshop
application/fastinfoset
application/fastsoap
application/fits
application/font-tdpfr pfr
application/h224
application/http
application/hyperstudio stk
application/iges
application/im-iscomposing+xml
application/index
application/index.cmd
application/index.obj
application/index.response
application/index.vnd
application/iotp
application/ipp
application/isup
application/javascript js
application/json json
application/kpml-request+xml
application/kpml-response+xml
application/mac-binhex40 hqx
application/mac-compactpro cpt
application/macwriteii
application/marc mrc
application/mathematica ma nb mb
application/mathml+xml mathml
application/mbms-associated-procedure-description+xml
application/mbms-deregister+xml
application/mbms-envelope+xml
application/mbms-msk+xml
application/mbms-msk-response+xml
application/mbms-protection-description+xml
application/mbms-reception-report+xml
application/mbms-register+xml
application/mbms-register-response+xml
application/mbms-user-service-description+xml
application/mbox mbox
application/mediaservercontrol+xml mscml
application/mikey
application/mp4 mp4s
application/mpeg4-generic
application/mpeg4-iod
application/mpeg4-iod-xmt
application/msword doc dot
application/mxf mxf
application/nasdata
application/news-message-id
application/news-transmission
application/nss
application/ocsp-request
application/ocsp-response
application/octet-stream bin dms lha lzh class so iso dmg dist distz pkg bpk dump elc
application/oda oda
application/oebps-package+xml
application/ogg ogg
application/parityfec
application/pdf pdf
application/pgp-encrypted pgp
application/pgp-keys
application/pgp-signature asc sig
application/pics-rules prf
application/pidf+xml
application/pkcs10 p10
application/pkcs7-mime p7m p7c
application/pkcs7-signature p7s
application/pkix-cert cer
application/pkix-crl crl
application/pkix-pkipath pkipath
application/pkixcmp pki
application/pls+xml pls
application/poc-settings+xml
application/postscript ai eps ps
application/prs.alvestrand.titrax-sheet
application/prs.cww cww
application/prs.nprend
application/prs.plucker
application/qsig
application/rdf+xml rdf
application/reginfo+xml rif
application/relax-ng-compact-syntax rnc
application/remote-printing
application/resource-lists+xml rl
application/riscos
application/rlmi+xml
application/rls-services+xml rs
application/rsd+xml rsd
application/rss+xml rss
application/rtf rtf
application/rtx
application/samlassertion+xml
application/samlmetadata+xml
application/sbml+xml sbml
application/sdp sdp
application/set-payment
application/set-payment-initiation setpay
application/set-registration
application/set-registration-initiation setreg
application/sgml
application/sgml-open-catalog
application/shf+xml shf
application/sieve
application/simple-filter+xml
application/simple-message-summary
application/simplesymbolcontainer
application/slate
application/smil
application/smil+xml smi smil
application/soap+fastinfoset
application/soap+xml
application/spirits-event+xml
application/srgs gram
application/srgs+xml grxml
application/ssml+xml ssml
application/timestamp-query
application/timestamp-reply
application/tve-trigger
application/vemmi
application/vividence.scriptfile
application/vnd.3gpp.bsf+xml
application/vnd.3gpp.pic-bw-large plb
application/vnd.3gpp.pic-bw-small psb
application/vnd.3gpp.pic-bw-var pvb
application/vnd.3gpp.sms
application/vnd.3gpp2.bcmcsinfo+xml
application/vnd.3gpp2.sms
application/vnd.3m.post-it-notes pwn
application/vnd.accpac.simply.aso aso
application/vnd.accpac.simply.imp imp
application/vnd.acucobol acu
application/vnd.acucorp atc acutc
application/vnd.adobe.xdp+xml xdp
application/vnd.adobe.xfdf xfdf
application/vnd.aether.imp
application/vnd.amiga.ami ami
application/vnd.anser-web-certificate-issue-initiation cii
application/vnd.anser-web-funds-transfer-initiation fti
application/vnd.antix.game-component atx
application/vnd.apple.installer+xml mpkg
application/vnd.audiograph aep
application/vnd.autopackage
application/vnd.avistar+xml
application/vnd.blueice.multipass mpm
application/vnd.bmi bmi
application/vnd.businessobjects rep
application/vnd.cab-jscript
application/vnd.canon-cpdl
application/vnd.canon-lips
application/vnd.cendio.thinlinc.clientconf
application/vnd.chemdraw+xml cdxml
application/vnd.chipnuts.karaoke-mmd mmd
application/vnd.cinderella cdy
application/vnd.cirpack.isdn-ext
application/vnd.claymore cla
application/vnd.clonk.c4group c4g c4d c4f c4p c4u
application/vnd.commerce-battelle
application/vnd.commonspace csp cst
application/vnd.contact.cmsg cdbcmsg
application/vnd.cosmocaller cmc
application/vnd.crick.clicker clkx
application/vnd.crick.clicker.keyboard clkk
application/vnd.crick.clicker.palette clkp
application/vnd.crick.clicker.template clkt
application/vnd.crick.clicker.wordbank clkw
application/vnd.criticaltools.wbs+xml wbs
application/vnd.ctc-posml pml
application/vnd.cups-pdf
application/vnd.cups-postscript
application/vnd.cups-ppd ppd
application/vnd.cups-raster
application/vnd.cups-raw
application/vnd.curl curl
application/vnd.cybank
application/vnd.data-vision.rdz rdz
application/vnd.denovo.fcselayout-link fe_launch
application/vnd.dna dna
application/vnd.dolby.mlp mlp
application/vnd.dpgraph dpg
application/vnd.dreamfactory dfac
application/vnd.dvb.esgcontainer
application/vnd.dvb.ipdcesgaccess
application/vnd.dxr
application/vnd.ecdis-update
application/vnd.ecowin.chart mag
application/vnd.ecowin.filerequest
application/vnd.ecowin.fileupdate
application/vnd.ecowin.series
application/vnd.ecowin.seriesrequest
application/vnd.ecowin.seriesupdate
application/vnd.enliven nml
application/vnd.epson.esf esf
application/vnd.epson.msf msf
application/vnd.epson.quickanime qam
application/vnd.epson.salt slt
application/vnd.epson.ssf ssf
application/vnd.ericsson.quickcall
application/vnd.eszigno3+xml es3 et3
application/vnd.eudora.data
application/vnd.ezpix-album ez2
application/vnd.ezpix-package ez3
application/vnd.fdf fdf
application/vnd.ffsns
application/vnd.fints
application/vnd.flographit gph
application/vnd.fluxtime.clip ftc
application/vnd.framemaker fm frame maker
application/vnd.frogans.fnc fnc
application/vnd.frogans.ltf ltf
application/vnd.fsc.weblaunch fsc
application/vnd.fujitsu.oasys oas
application/vnd.fujitsu.oasys2 oa2
application/vnd.fujitsu.oasys3 oa3
application/vnd.fujitsu.oasysgp fg5
application/vnd.fujitsu.oasysprs bh2
application/vnd.fujixerox.art-ex
application/vnd.fujixerox.art4
application/vnd.fujixerox.hbpl
application/vnd.fujixerox.ddd ddd
application/vnd.fujixerox.docuworks xdw
application/vnd.fujixerox.docuworks.binder xbd
application/vnd.fut-misnet
application/vnd.fuzzysheet fzs
application/vnd.genomatix.tuxedo txd
application/vnd.google-earth.kml+xml kml
application/vnd.google-earth.kmz kmz
application/vnd.grafeq gqf gqs
application/vnd.gridmp
application/vnd.groove-account gac
application/vnd.groove-help ghf
application/vnd.groove-identity-message gim
application/vnd.groove-injector grv
application/vnd.groove-tool-message gtm
application/vnd.groove-tool-template tpl
application/vnd.groove-vcard vcg
application/vnd.handheld-entertainment+xml zmm
application/vnd.hbci hbci
application/vnd.hcl-bireports
application/vnd.hhe.lesson-player les
application/vnd.hp-hpgl hpgl
application/vnd.hp-hpid hpid
application/vnd.hp-hps hps
application/vnd.hp-jlyt jlt
application/vnd.hp-pcl pcl
application/vnd.hp-pclxl pclxl
application/vnd.httphone
application/vnd.hzn-3d-crossword x3d
application/vnd.ibm.afplinedata
application/vnd.ibm.electronic-media
application/vnd.ibm.minipay mpy
application/vnd.ibm.modcap afp listafp list3820
application/vnd.ibm.rights-management irm
application/vnd.ibm.secure-container sc
application/vnd.igloader igl
application/vnd.immervision-ivp ivp
application/vnd.immervision-ivu ivu
application/vnd.informedcontrol.rms+xml
application/vnd.intercon.formnet xpw xpx
application/vnd.intertrust.digibox
application/vnd.intertrust.nncp
application/vnd.intu.qbo qbo
application/vnd.intu.qfx qfx
application/vnd.ipunplugged.rcprofile rcprofile
application/vnd.irepository.package+xml irp
application/vnd.is-xpr xpr
application/vnd.jam jam
application/vnd.japannet-directory-service
application/vnd.japannet-jpnstore-wakeup
application/vnd.japannet-payment-wakeup
application/vnd.japannet-registration
application/vnd.japannet-registration-wakeup
application/vnd.japannet-setstore-wakeup
application/vnd.japannet-verification
application/vnd.japannet-verification-wakeup
application/vnd.jcp.javame.midlet-rms rms
application/vnd.jisp jisp
application/vnd.kahootz ktz ktr
application/vnd.kde.karbon karbon
application/vnd.kde.kchart chrt
application/vnd.kde.kformula kfo
application/vnd.kde.kivio flw
application/vnd.kde.kontour kon
application/vnd.kde.kpresenter kpr kpt
application/vnd.kde.kspread ksp
application/vnd.kde.kword kwd kwt
application/vnd.kenameaapp htke
application/vnd.kidspiration kia
application/vnd.kinar kne knp
application/vnd.koan skp skd skt skm
application/vnd.liberty-request+xml
application/vnd.llamagraphics.life-balance.desktop lbd
application/vnd.llamagraphics.life-balance.exchange+xml lbe
application/vnd.lotus-1-2-3 123
application/vnd.lotus-approach apr
application/vnd.lotus-freelance pre
application/vnd.lotus-notes nsf
application/vnd.lotus-organizer org
application/vnd.lotus-screencam scm
application/vnd.lotus-wordpro lwp
application/vnd.macports.portpkg portpkg
application/vnd.marlin.drm.actiontoken+xml
application/vnd.marlin.drm.conftoken+xml
application/vnd.marlin.drm.mdcf
application/vnd.mcd mcd
application/vnd.medcalcdata mc1
application/vnd.mediastation.cdkey cdkey
application/vnd.meridian-slingshot
application/vnd.mfer mwf
application/vnd.mfmp mfm
application/vnd.micrografx.flo flo
application/vnd.micrografx.igx igx
application/vnd.mif mif
application/vnd.minisoft-hp3000-save
application/vnd.mitsubishi.misty-guard.trustweb
application/vnd.mobius.daf daf
application/vnd.mobius.dis dis
application/vnd.mobius.mbk mbk
application/vnd.mobius.mqy mqy
application/vnd.mobius.msl msl
application/vnd.mobius.plc plc
application/vnd.mobius.txf txf
application/vnd.mophun.application mpn
application/vnd.mophun.certificate mpc
application/vnd.motorola.flexsuite
application/vnd.motorola.flexsuite.adsi
application/vnd.motorola.flexsuite.fis
application/vnd.motorola.flexsuite.gotap
application/vnd.motorola.flexsuite.kmr
application/vnd.motorola.flexsuite.ttc
application/vnd.motorola.flexsuite.wem
application/vnd.mozilla.xul+xml xul
application/vnd.ms-artgalry cil
application/vnd.ms-asf asf
application/vnd.ms-cab-compressed cab
application/vnd.ms-excel xls xlm xla xlc xlt xlw
application/vnd.ms-fontobject eot
application/vnd.ms-htmlhelp chm
application/vnd.ms-ims ims
application/vnd.ms-lrm lrm
application/vnd.ms-playready.initiator+xml
application/vnd.ms-powerpoint ppt pps pot
application/vnd.ms-project mpp mpt
application/vnd.ms-tnef
application/vnd.ms-wmdrm.lic-chlg-req
application/vnd.ms-wmdrm.lic-resp
application/vnd.ms-wmdrm.meter-chlg-req
application/vnd.ms-wmdrm.meter-resp
application/vnd.ms-works wps wks wcm wdb
application/vnd.ms-wpl wpl
application/vnd.ms-xpsdocument xps
application/vnd.mseq mseq
application/vnd.msign
application/vnd.music-niff
application/vnd.musician mus
application/vnd.ncd.control
application/vnd.nervana
application/vnd.netfpx
application/vnd.neurolanguage.nlu nlu
application/vnd.noblenet-directory nnd
application/vnd.noblenet-sealer nns
application/vnd.noblenet-web nnw
application/vnd.nokia.catalogs
application/vnd.nokia.conml+wbxml
application/vnd.nokia.conml+xml
application/vnd.nokia.isds-radio-presets
application/vnd.nokia.iptv.config+xml
application/vnd.nokia.landmark+wbxml
application/vnd.nokia.landmark+xml
application/vnd.nokia.landmarkcollection+xml
application/vnd.nokia.n-gage.ac+xml
application/vnd.nokia.n-gage.data ngdat
application/vnd.nokia.n-gage.symbian.install n-gage
application/vnd.nokia.ncd
application/vnd.nokia.pcd+wbxml
application/vnd.nokia.pcd+xml
application/vnd.nokia.radio-preset rpst
application/vnd.nokia.radio-presets rpss
application/vnd.novadigm.edm edm
application/vnd.novadigm.edx edx
application/vnd.novadigm.ext ext
application/vnd.oasis.opendocument.chart odc
application/vnd.oasis.opendocument.chart-template otc
application/vnd.oasis.opendocument.formula odf
application/vnd.oasis.opendocument.formula-template otf
application/vnd.oasis.opendocument.graphics odg
application/vnd.oasis.opendocument.graphics-template otg
application/vnd.oasis.opendocument.image odi
application/vnd.oasis.opendocument.image-template oti
application/vnd.oasis.opendocument.presentation odp
application/vnd.oasis.opendocument.presentation-template otp
application/vnd.oasis.opendocument.spreadsheet ods
application/vnd.oasis.opendocument.spreadsheet-template ots
application/vnd.oasis.opendocument.text odt
application/vnd.oasis.opendocument.text-master otm
application/vnd.oasis.opendocument.text-template ott
application/vnd.oasis.opendocument.text-web oth
application/vnd.obn
application/vnd.olpc-sugar xo
application/vnd.oma-scws-config
application/vnd.oma-scws-http-request
application/vnd.oma-scws-http-response
application/vnd.oma.bcast.associated-procedure-parameter+xml
application/vnd.oma.bcast.drm-trigger+xml
application/vnd.oma.bcast.imd+xml
application/vnd.oma.bcast.notification+xml
application/vnd.oma.bcast.sgboot
application/vnd.oma.bcast.sgdd+xml
application/vnd.oma.bcast.sgdu
application/vnd.oma.bcast.simple-symbol-container
application/vnd.oma.bcast.smartcard-trigger+xml
application/vnd.oma.bcast.sprov+xml
application/vnd.oma.dd2+xml dd2
application/vnd.oma.drm.risd+xml
application/vnd.oma.group-usage-list+xml
application/vnd.oma.poc.groups+xml
application/vnd.oma.xcap-directory+xml
application/vnd.omads-email+xml
application/vnd.omads-file+xml
application/vnd.omads-folder+xml
application/vnd.omaloc-supl-init
application/vnd.openofficeorg.extension oxt
application/vnd.osa.netdeploy
application/vnd.osgi.dp dp
application/vnd.otps.ct-kip+xml
application/vnd.palm prc pdb pqa oprc
application/vnd.paos.xml
application/vnd.pg.format str
application/vnd.pg.osasli ei6
application/vnd.piaccess.application-licence
application/vnd.picsel efif
application/vnd.poc.group-advertisement+xml
application/vnd.pocketlearn plf
application/vnd.powerbuilder6 pbd
application/vnd.powerbuilder6-s
application/vnd.powerbuilder7
application/vnd.powerbuilder7-s
application/vnd.powerbuilder75
application/vnd.powerbuilder75-s
application/vnd.preminet
application/vnd.previewsystems.box box
application/vnd.proteus.magazine mgz
application/vnd.publishare-delta-tree qps
application/vnd.pvi.ptid1 ptid
application/vnd.pwg-multiplexed
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application/vnd.yamaha.smaf-audio saf
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application/watcherinfo+xml
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application/wspolicy+xml wspolicy
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application/x-director dcr dir dxr fgd
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application/xmpp+xml
application/xop+xml xop
application/xslt+xml xslt
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image/vnd.fujixerox.edmics-mmr mmr
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image/vnd.globalgraphics.pgb
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image/vnd.mix
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image/vnd.net-fpx npx
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x-conference/x-cooltalk ice

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<profile name="internal">
<!--
This is a sofia sip profile/user agent. This will service exactly one ip and port.
In FreeSWITCH you can run multiple sip user agents on their own ip and port.
When you hear someone say "sofia profile" this is what they are talking about.
-->
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!--aliases are other names that will work as a valid profile name for this profile-->
<aliases>
<!--
<alias name="default"/>
-->
</aliases>
<!-- Outbound Registrations -->
<domains>
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
<domain name="all" alias="true" parse="false"/>
</domains>
<settings>
<!-- inject delay between dtmf digits on send to help some slow interpreters (also per channel with rtp_digit_delay var -->
<!-- <param name="rtp-digit-delay" value="40"/>-->
<!--
When calls are in no media this will bring them back to media
when you press the hold button.
-->
<!--<param name="media-option" value="resume-media-on-hold"/> -->
<!--
This will allow a call after an attended transfer go back to
bypass media after an attended transfer.
-->
<!--<param name="media-option" value="bypass-media-after-att-xfer"/>-->
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<!-- If you want FreeSWITCH to shutdown if this profile fails to load, uncomment the next line. -->
<!-- <param name="shutdown-on-fail" value="true"/> -->
<param name="sip-trace" value="no"/>
<param name="sip-capture" value="no"/>
<!-- Use presence_map.conf.xml to convert extension regex to presence protos for routing -->
<!-- <param name="presence-proto-lookup" value="true"/> -->
<!-- Don't be picky about negotiated DTMF just always offer 2833 and accept both 2833 and INFO -->
<param name="liberal-dtmf" value="true"/>
<!--
Sometimes, in extremely rare edge cases, the Sofia SIP stack may stop
responding. These options allow you to enable and control a watchdog
on the Sofia SIP stack so that if it stops responding for the
specified number of milliseconds, it will cause FreeSWITCH to crash
immediately. This is useful if you run in an HA environment and
need to ensure automated recovery from such a condition. Note that if
your server is idle a lot, the watchdog may fire due to not receiving
any SIP messages. Thus, if you expect your system to be idle, you
should leave the watchdog disabled. It can be toggled on and off
through the FreeSWITCH CLI either on an individual profile basis or
globally for all profiles. So, if you run in an HA environment with a
master and slave, you should use the CLI to make sure the watchdog is
only enabled on the master.
If such crash occurs, FreeSWITCH will dump core if allowed. The
stacktrace will include function watchdog_triggered_abort().
-->
<param name="watchdog-enabled" value="no"/>
<param name="watchdog-step-timeout" value="30000"/>
<param name="watchdog-event-timeout" value="30000"/>
<param name="log-auth-failures" value="false"/>
<param name="forward-unsolicited-mwi-notify" value="false"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="5069"/> <!-- FIXME: ${SIPCON_REMOTE_PORT} -->
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="rtp-timer-name" value="soft"/>
<!-- ip address to use for rtp, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="sip-ip" value="127.0.0.1"/> <!-- FIXME: ${SIPCON_REMOTE} -->
<!--<param name="hold-music" value="$${hold_music}"/>-->
<param name="apply-nat-acl" value="nat.auto"/>
<!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
<!-- <param name="cid-in-1xx" value="false"/> -->
<!-- extended info parsing -->
<!-- <param name="extended-info-parsing" value="true"/> -->
<param name="aggressive-nat-detection" value="false"/>
<!--
There are known issues (asserts and segfaults) when 100rel is enabled.
It is not recommended to enable 100rel at this time.
-->
<!--<param name="enable-100rel" value="true"/>-->
<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
<!-- RFC3263 Section 4.3 -->
<!--<param name="disable-srv503" value="true"/>-->
<!-- Enable Compact SIP headers. -->
<!--<param name="enable-compact-headers" value="true"/>-->
<!--
enable/disable session timers
-->
<!--<param name="enable-timer" value="false"/>-->
<!--<param name="minimum-session-expires" value="120"/>-->
<param name="apply-inbound-acl" value="domains"/>
<!--
This defines your local network, by default we detect your local network
and create this localnet.auto ACL for this.
-->
<param name="local-network-acl" value="localnet.auto"/>
<param name="apply-register-acl" value="domains"/>
<param name="apply-candidate-acl" value="domains"/>
<!--<param name="dtmf-type" value="info"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-message-query-on-register" value="true"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-presence-on-register" value="first-only"/> -->
<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
<!-- Remote-Party-ID header -->
<!--<param name="caller-id-type" value="rpid"/>-->
<!-- P-*-Identity family of headers -->
<!--<param name="caller-id-type" value="pid"/>-->
<!-- neither one -->
<!--<param name="caller-id-type" value="none"/>-->
<param name="record-path" value="$${recordings_dir}"/>
<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<!--enable to use presence -->
<param name="manage-presence" value="false"/>
<!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
<!--<param name="presence-probe-on-register" value="true"/>-->
<!--<param name="manage-shared-appearance" value="true"/>-->
<!-- used to share presence info across sofia profiles -->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
<param name="presence-privacy" value="$${presence_privacy}"/>
<!-- ************************************************* -->
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<!--<param name="max-proceeding" value="1000"/>-->
<!--session timers for all call to expire after the specified seconds -->
<!--<param name="session-timeout" value="1800"/>-->
<!-- Can be 'true' or 'contact' -->
<!--<param name="multiple-registrations" value="contact"/>-->
<!--set to 'greedy' if you want your codec list to take precedence -->
<param name="inbound-codec-negotiation" value="generous"/>
<!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--<param name="unregister-on-options-fail" value="true"/>-->
<!-- Send an OPTIONS packet to all registered endpoints -->
<!--<param name="all-reg-options-ping" value="true"/>-->
<!-- Send an OPTIONS packet to NATed registered endpoints. Can be 'true' or 'udp-only'. -->
<!--<param name="nat-options-ping" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${internal_ssl_enable}"/>
<!-- Set to true to not bind on the normal sip-port but only on the TLS port -->
<param name="tls-only" value="false"/>
<!-- additional bind parameters for TLS -->
<param name="tls-bind-params" value="transport=tls"/>
<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
<param name="tls-sip-port" value="$${internal_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<!--<param name="tls-cert-dir" value=""/>-->
<!-- Optionally set the passphrase password used by openSSL to encrypt/decrypt TLS private key files -->
<param name="tls-passphrase" value=""/>
<!-- Verify the date on TLS certificates -->
<param name="tls-verify-date" value="true"/>
<!-- TLS verify policy, when registering/inviting gateways with other servers (outbound) or handling inbound registration/invite requests how should we verify their certificate -->
<!-- set to 'in' to only verify incoming connections, 'out' to only verify outgoing connections, 'all' to verify all connections, also 'in_subjects', 'out_subjects' and 'all_subjects' for subject validation. Multiple policies can be split with a '|' pipe -->
<param name="tls-verify-policy" value="none"/>
<!-- Certificate max verify depth to use for validating peer TLS certificates when the verify policy is not none -->
<param name="tls-verify-depth" value="2"/>
<!-- If the tls-verify-policy is set to subjects_all or subjects_in this sets which subjects are allowed, multiple subjects can be split with a '|' pipe -->
<param name="tls-verify-in-subjects" value=""/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!-- turn on auto-flush during bridge (skip timer sleep when the socket already has data)
(reduces delay on latent connections default true, must be disabled explicitly)-->
<!--<param name="rtp-autoflush-during-bridge" value="false"/>-->
<!--If you don't want to pass through timestamps from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!-- Or, if you have PGSQL support, you can use that -->
<!--<param name="odbc-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='' options='-c client_min_messages=NOTICE' application_name='freeswitch'" />-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-media" value="true"/>-->
<!-- Let calls hit the dialplan before selecting codec for the a-leg -->
<param name="inbound-late-negotiation" value="false"/>
<!-- Allow ZRTP clients to negotiate end-to-end security associations (also enables late negotiation) -->
<param name="inbound-zrtp-passthru" value="true"/>
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
<!-- <param name="accept-blind-reg" value="true"/> -->
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<!-- <param name="accept-blind-auth" value="true"/> -->
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
<!-- <param name="suppress-cng" value="true"/> -->
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!-- Handle 302 Redirect in the dialplan -->
<!--<param name="manual-redirect" value="true"/> -->
<!-- Disable Transfer -->
<!--<param name="disable-transfer" value="true"/> -->
<!-- Disable Register -->
<!--<param name="disable-register" value="true"/> -->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
<param name="auth-calls" value="$${internal_auth_calls}"/>
<!-- Force the user and auth-user to match. -->
<param name="inbound-reg-force-matching-username" value="true"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- external_sip_ip
Used as the public IP address for SDP.
Can be an one of:
ip address - "12.34.56.78"
a stun server lookup - "stun:stun.server.com"
a DNS name - "host:host.server.com"
auto - Use guessed ip.
auto-nat - Use ip learned from NAT-PMP or UPNP
-->
<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
<param name="ext-sip-ip" value="$${local_ip_v4}"/>
<!-- rtp inactivity timeout -->
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
<!--
These are enabled to make the default config work better out of the box.
If you need more than ONE domain you'll need to not use these options.
-->
<!--all inbound reg will look in this domain for the users -->
<param name="force-register-domain" value="$${domain}"/>
<!--force the domain in subscriptions to this value -->
<param name="force-subscription-domain" value="$${domain}"/>
<!--all inbound reg will stored in the db using this domain -->
<param name="force-register-db-domain" value="$${domain}"/>
<!-- uncomment for sip over websocket support -->
<!--<param name="ws-binding" value=":5066"/>-->
<!-- uncomment for sip over secure websocket support -->
<!-- You need wss.pem in /usr/local/freeswitch/certs for wss -->
<!--<param name="wss-binding" value=":7443"/>-->
<!--<param name="delete-subs-on-register" value="false"/>-->
<!-- launch a new thread to process each new inbound register when using heavier backends -->
<!-- <param name="inbound-reg-in-new-thread" value="true"/> -->
<!-- enable rtcp on every channel also can be done per leg basis with rtcp_audio_interval_msec variable set to passthru to pass it across a call-->
<!--<param name="rtcp-audio-interval-msec" value="5000"/>-->
<!--<param name="rtcp-video-interval-msec" value="5000"/>-->
<!--force suscription expires to a lower value than requested-->
<!--<param name="force-subscription-expires" value="60"/>-->
<!-- add a random deviation to the expires value of the 202 Accepted -->
<!--<param name="sip-subscription-max-deviation" value="120"/>-->
<!-- disable register and transfer which may be undesirable in a public switch -->
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!--
enable-3pcc can be set to either 'true' or 'proxy', true accepts the call
right away, proxy waits until the call has been answered then sends accepts
-->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use at your own risk or if you know what this does.-->
<!--<param name="NDLB-force-rport" value="true"/>-->
<!--
Choose the realm challenge key. Default is auto_to if not set.
auto_from - uses the from field as the value for the sip realm.
auto_to - uses the to field as the value for the sip realm.
<anyvalue> - you can input any value to use for the sip realm.
If you want URL dialing to work you'll want to set this to auto_from.
If you use any other value besides auto_to or auto_from you'll
loose the ability to do multiple domains.
Note: comment out to restore the behavior before 2008-09-29
-->
<param name="challenge-realm" value="auto_from"/>
<!--<param name="disable-rtp-auto-adjust" value="true"/>-->
<!-- on inbound calls make the uuid of the session equal to the sip call id of that call -->
<!--<param name="inbound-use-callid-as-uuid" value="true"/>-->
<!-- on outbound calls set the callid to match the uuid of the session -->
<!--<param name="outbound-use-uuid-as-callid" value="true"/>-->
<!-- set to false disable this feature -->
<!--<param name="rtp-autofix-timing" value="false"/>-->
<!-- set this param to false if your gateway for some reason hates X- headers that it is supposed to ignore-->
<!--<param name="pass-callee-id" value="false"/>-->
<!-- clear clears them all or supply the name to add or the name
prefixed with ~ to remove valid values:
clear
CISCO_SKIP_MARK_BIT_2833
SONUS_SEND_INVALID_TIMESTAMP_2833
-->
<!--<param name="auto-rtp-bugs" data="clear"/>-->
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
<!--<param name="disable-srv" value="false" />-->
<!--<param name="disable-naptr" value="false" />-->
<!-- The following can be used to fine-tune timers within sofia's transport layer
Those settings are for advanced users and can safely be left as-is -->
<!-- Initial retransmission interval (in milliseconds).
Set the T1 retransmission interval used by the SIP transaction engine.
The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. -->
<!-- <param name="timer-T1" value="500" /> -->
<!-- Transaction timeout (defaults to T1 * 64).
Set the T1x64 timeout value used by the SIP transaction engine.
The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine.
The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. -->
<!-- <param name="timer-T1X64" value="32000" /> -->
<!-- Maximum retransmission interval (in milliseconds).
Set the maximum retransmission interval used by the SIP transaction engine.
The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine.
Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially
until the timer B fires. -->
<!-- <param name="timer-T2" value="4000" /> -->
<!--
Transaction lifetime (in milliseconds).
Set the lifetime for completed transactions used by the SIP transaction engine.
A completed transaction is kept around for the duration of T4 in order to catch late responses.
The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. -->
<!-- <param name="timer-T4" value="4000" /> -->
<!-- Turn on a jitterbuffer for every call -->
<!-- <param name="auto-jitterbuffer-msec" value="60"/> -->
<!-- By default mod_sofia will ignore the codecs in the sdp for hold/unhold operations
Set this to true if you want to actually parse the sdp and re-negotiate the codec during hold/unhold.
It's probably not what you want so stick with the default unless you really need to change this.
-->
<!--<param name="renegotiate-codec-on-hold" value="true"/>-->
</settings>
</profile>

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@ -0,0 +1,448 @@
<include>
<!-- Preprocessor Variables
These are introduced when configuration strings must be consistent across modules.
NOTICE: YOU CAN NOT COMMENT OUT AN X-PRE-PROCESS line, Remove the line instead.
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
-->
<X-PRE-PROCESS cmd="set" data="default_password=1234"/>
<!-- Did you change it yet? -->
<!--
The following variables are set dynamically - calculated if possible by freeswitch - and
are available to the config as $${variable}. You can see their calculated value via fs_cli
by entering eval $${variable}
hostname
local_ip_v4
local_mask_v4
local_ip_v6
switch_serial
base_dir
recordings_dir
sound_prefix
sounds_dir
conf_dir
log_dir
run_dir
db_dir
mod_dir
htdocs_dir
script_dir
temp_dir
grammar_dir
certs_dir
storage_dir
cache_dir
core_uuid
zrtp_enabled
nat_public_addr
nat_private_addr
nat_type
-->
<X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/>
<!--<Z-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/allison"/> -->
<!--
This setting is what sets the default domain FreeSWITCH will use if all else fails.
FreeSWICH will default to $${local_ip_v4} unless changed. Changing this setting does
affect the sip authentication. Please review conf/directory/default.xml for more
information on this topic.
-->
<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
<X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>
<X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/>
<!-- <X-PRE-PROCESS cmd="set" data="use_profile=external"/> -->
<X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH"/>
<!--
Enable ZRTP globally you can override this on a per channel basis
http://wiki.freeswitch.org/wiki/ZRTP (on how to enable zrtp)
-->
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
<!--
NOTICE: When using SRTP it's critical that you do not offer or accept
variable bit rate codecs, doing so would leak information and possibly
compromise your SRTP stream. (FS-6404)
Supported SRTP Crypto Suites:
AEAD_AES_256_GCM_8
____________________________________________________________________________
This algorithm is identical to AEAD_AES_256_GCM (see Section 5.2 of
[RFC5116]), except that the tag length, t, is 8, and an
authentication tag with a length of 8 octets (64 bits) is used.
An AEAD_AES_256_GCM_8 ciphertext is exactly 8 octets longer than its
corresponding plaintext.
AEAD_AES_128_GCM_8
____________________________________________________________________________
This algorithm is identical to AEAD_AES_128_GCM (see Section 5.1 of
[RFC5116]), except that the tag length, t, is 8, and an
authentication tag with a length of 8 octets (64 bits) is used.
An AEAD_AES_128_GCM_8 ciphertext is exactly 8 octets longer than its
corresponding plaintext.
AES_CM_256_HMAC_SHA1_80 | AES_CM_192_HMAC_SHA1_80 | AES_CM_128_HMAC_SHA1_80
____________________________________________________________________________
AES_CM_128_HMAC_SHA1_80 is the SRTP default AES Counter Mode cipher
and HMAC-SHA1 message authentication with an 80-bit authentication
tag. The master-key length is 128 bits and has a default lifetime of
a maximum of 2^48 SRTP packets or 2^31 SRTCP packets, whichever comes
first.
AES_CM_256_HMAC_SHA1_32 | AES_CM_192_HMAC_SHA1_32 | AES_CM_128_HMAC_SHA1_32
____________________________________________________________________________
This crypto-suite is identical to AES_CM_128_HMAC_SHA1_80 except that
the authentication tag is 32 bits. The length of the base64-decoded key and
salt value for this crypto-suite MUST be 30 octets i.e., 240 bits; otherwise,
the crypto attribute is considered invalid.
AES_CM_128_NULL_AUTH
____________________________________________________________________________
The SRTP default cipher (AES-128 Counter Mode), but to use no authentication
method. This policy is NOT RECOMMENDED unless it is unavoidable; see
Section 7.5 of [RFC3711].
SRTP variables that modify behaviors based on direction/leg:
rtp_secure_media
____________________________________________________________________________
possible values:
mandatory - Accept/Offer SAVP negotiation ONLY
optional - Accept/Offer SAVP/AVP with SAVP preferred
forbidden - More useful for inbound to deny SAVP negotiation
false - implies forbidden
true - implies mandatory
default if not set is accept SAVP inbound if offered.
rtp_secure_media_inbound | rtp_secure_media_outbound
____________________________________________________________________________
This is the same as rtp_secure_media, but would apply to either inbound
or outbound offers specifically.
How to specify crypto suites:
____________________________________________________________________________
By default without specifying any crypto suites FreeSWITCH will offer
crypto suites from strongest to weakest accepting the strongest each
endpoint has in common. If you wish to force specific crypto suites you
can do so by appending the suites in a comma separated list in the order
that you wish to offer them in.
Examples:
rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80
Additionally you can narrow this down on either inbound or outbound by
specifying as so:
rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8
rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8
rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8
rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8
rtp_secure_media_suites
____________________________________________________________________________
Optionaly you can use rtp_secure_media_suites to dictate the suite list
and only use rtp_secure_media=[optional|mandatory|false|true] without having
to dictate the suite list with the rtp_secure_media* variables.
-->
<!--
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 their purpose.
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
-->
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,AMR,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,AMR,GSM"/>
<!--
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
-->
<X-PRE-PROCESS cmd="set" data="xmpp_client_profile=xmppc"/>
<X-PRE-PROCESS cmd="set" data="xmpp_server_profile=xmpps"/>
<!--
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
-->
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<!-- 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
-->
<X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
<!-- 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
-->
<X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/>
<!-- unroll-loops
Used to turn on sip loopback unrolling.
-->
<X-PRE-PROCESS cmd="set" data="unroll_loops=true"/>
<!-- 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.
-->
<X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
<X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
<!-- various debug and defaults -->
<X-PRE-PROCESS cmd="set" data="call_debug=false"/>
<X-PRE-PROCESS cmd="set" data="console_loglevel=info"/>
<X-PRE-PROCESS cmd="set" data="default_areacode=918"/>
<X-PRE-PROCESS cmd="set" data="default_country=US"/>
<!-- if false or undefined, the destination number is included in presence NOTIFY dm:note.
if true, the destination number is not included -->
<X-PRE-PROCESS cmd="set" data="presence_privacy=false"/>
<X-PRE-PROCESS cmd="set" data="au-ring=%(400,200,383,417);%(400,2000,383,417)"/>
<X-PRE-PROCESS cmd="set" data="be-ring=%(1000,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="ca-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="cn-ring=%(1000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="cy-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="cz-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="de-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dk-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="dz-ring=%(1500,3500,425)"/>
<X-PRE-PROCESS cmd="set" data="eg-ring=%(2000,1000,475,375)"/>
<X-PRE-PROCESS cmd="set" data="es-ring=%(1500,3000,425)"/>
<X-PRE-PROCESS cmd="set" data="fi-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="fr-ring=%(1500,3500,440)"/>
<X-PRE-PROCESS cmd="set" data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="hu-ring=%(1250,3750,425)"/>
<X-PRE-PROCESS cmd="set" data="il-ring=%(1000,3000,400)"/>
<X-PRE-PROCESS cmd="set" data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/>
<X-PRE-PROCESS cmd="set" data="jp-ring=%(1000,2000,420,380)"/>
<X-PRE-PROCESS cmd="set" data="ko-ring=%(1000,2000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="pk-ring=%(1000,2000,400)"/>
<X-PRE-PROCESS cmd="set" data="pl-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ro-ring=%(1850,4150,475,425)"/>
<X-PRE-PROCESS cmd="set" data="rs-ring=%(1000,4000,425)"/>
<X-PRE-PROCESS cmd="set" data="ru-ring=%(800,3200,425)"/>
<X-PRE-PROCESS cmd="set" data="sa-ring=%(1200,4600,425)"/>
<X-PRE-PROCESS cmd="set" data="tr-ring=%(2000,4000,450)"/>
<X-PRE-PROCESS cmd="set" data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/>
<X-PRE-PROCESS cmd="set" data="us-ring=%(2000,4000,440,480)"/>
<X-PRE-PROCESS cmd="set" data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/>
<X-PRE-PROCESS cmd="set" data="beep=%(1000,0,640)"/>
<X-PRE-PROCESS cmd="set" data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/>
<!--
Digits Dialed filter: (FS-6940)
The digits stream may contain valid credit card numbers or social security numbers, These digit
filters will allow you to make a valant effort to stamp out sensitive information for
PCI/HIPPA compliance. (see xml_cdr dialed_digits)
df_us_ssn = US Social Security Number pattern
df_us_luhn = Visa, MasterCard, American Express, Diners Club, Discover and JCB
-->
<X-PRE-PROCESS cmd="set" data="df_us_ssn=(?!219099999|078051120)(?!666|000|9\d{2})\d{3}(?!00)\d{2}(?!0{4})\d{4}"/>
<X-PRE-PROCESS cmd="set" data="df_luhn=?:4[0-9]{12}(?:[0-9]{3})?|5[1-5][0-9]{14}|3[47][0-9]{13}|3(?:0[0-5]|[68][0-9])[0-9]{11}|6(?:011|5[0-9]{2})[0-9]{12}|(?:2131|1800|35\d{3})\d{11}"/>
<!-- change XX to X below to enable -->
<XX-PRE-PROCESS cmd="set" data="digits_dialed_filter=(($${df_luhn})|($${df_us_ssn}))"/>
<!--
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
-->
<X-PRE-PROCESS cmd="set" data="default_provider=example.com"/>
<X-PRE-PROCESS cmd="set" data="default_provider_username=joeuser"/>
<X-PRE-PROCESS cmd="set" data="default_provider_password=password"/>
<X-PRE-PROCESS cmd="set" data="default_provider_from_domain=example.com"/>
<!-- true or false -->
<X-PRE-PROCESS cmd="set" data="default_provider_register=false"/>
<X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
<!--
SIP and TLS settings. http://wiki.freeswitch.org/wiki/Tls
valid options: sslv2,sslv3,sslv23,tlsv1,tlsv1.1,tlsv1.2
default: tlsv1,tlsv1.1,tlsv1.2
-->
<X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/>
<!--
TLS cipher suite: default ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH
The actual ciphers supported will change per platform.
openssl ciphers -v 'ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH'
Will show you what is available in your verion of openssl.
-->
<X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>
<!-- Internal SIP Profile -->
<X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
<X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
<X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
<X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
<!-- External SIP Profile -->
<X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
<X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
<X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
<X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>
<!-- Video Settings -->
<!-- Setting the max bandwdith -->
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_in=3mb"/>
<X-PRE-PROCESS cmd="set" data="rtp_video_max_bandwidth_out=3mb"/>
<!-- WebRTC Video -->
<!-- Suppress CNG for WebRTC Audio -->
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
<!-- Enable liberal DTMF for those that can't get it right -->
<X-PRE-PROCESS cmd="set" data="rtp_liberal_dtmf=true"/>
<!-- Helps with WebRTC Audio -->
<!-- Stock Video Avatars -->
<X-PRE-PROCESS cmd="set" data="video_mute_png=$${images_dir}/default-mute.png"/>
<X-PRE-PROCESS cmd="set" data="video_no_avatar_png=$${images_dir}/default-avatar.png"/>
</include>

View File

@ -0,0 +1,24 @@
#!/bin/sh
NETDIR="$(cd $(dirname "$0")/..; pwd)"
(sleep 5; echo; echo; echo "NOTE: type 'shutdown' to quit"; echo) &
set -ex
export PYTHONPATH="$NETDIR/freeswitch/python:$PYTHONPATH"
if ! [ -d /usr/lib/freeswitch/mod ]; then
echo "ERROR: missing dir /usr/lib/freeswitch/mod"
exit 1
fi
if ! [ -e freeswitch/mod ]; then
ln -sf /usr/lib/freeswitch/mod freeswitch/mod
fi
freeswitch \
-nf \
-nonat \
-nonatmap \
-nocal \
-nort \
-c \
-base ./freeswitch

View File

@ -0,0 +1,53 @@
import json
import os
import subprocess
script_dir = os.path.dirname(__file__)
timeout_ms = 2000
def query_mslookup(query_str):
result_line = subprocess.check_output([
'osmo-mslookup-client', query_str, '-f', 'json'])
if isinstance(result_line, bytes):
result_line = result_line.decode('ascii')
return json.loads(result_line)
def query_mslookup_msisdn(msisdn):
return query_mslookup('sip.voice.%s.msisdn' % str(msisdn))
def handler(session, args):
print('[dialplan-dgsm] handler')
msisdn = session.getVariable('destination_number')
print('[dialplan-dgsm] resolving: sip.voice.' + str(msisdn) + '.msisdn')
# Run mslookup.py. In theory, we should be able to import it and call mslookup.resolve() directly, however this
# has lead to hard-to-debug segfaults when calling it the second time. For now, use a separate python process.
try:
result = query_mslookup_msisdn(msisdn)
print('[dialplan-dgsm] result: ' + str(result))
# This example only makes use of IPv4
if not result['v4']:
print('[dialplan-dgsm] no IPv4 result from mslookup')
session.hangup('UNALLOCATED_NUMBER')
return
sip_ip, sip_port = result['v4'] # osmo-dev defaults: same as ${SIPCON_LOCAL}
dial_str = 'sofia/internal/sip:{}@{}:{}'.format(msisdn, sip_ip, sip_port)
print('[dialplan-dgsm] dial_str: ' + str(dial_str))
session.execute('bridge', dial_str)
except:
print('[dialplan-dgsm]: could not resolve MSISDN {} with mslookup.py'.format(msisdn))
session.hangup('UNALLOCATED_NUMBER')
# Freeswitch refuses to load the module without this
def fsapi(session, stream, env, args):
stream.write(env.serialize())
if __name__ == '__main__':
import json
import sys
print(json.dumps(query_mslookup(sys.argv[1])))

View File

@ -86,14 +86,32 @@ bsc="LD_LIBRARY_PATH=/usr/local/lib gdb -ex run --args osmo-bsc -c osmo-bsc.cfg"
if [ "x${MSC_MNCC}" != "xinternal" ]; then
sipcon="osmo-sip-connector -c osmo-sip-connector.cfg"
case "${SIPCON_SERVER}" in
"kamailio")
# Require kamailio (PATH hack is needed for Debian)
kamailio="$(PATH="$PATH:/usr/sbin:/sbin" which kamailio)"
if [ -z "$kamailio" ]; then
echo "ERROR: kamailio is not installed, but it's required for external MNCC."
echo "ERROR: kamailio is not installed."
echo "After installing it, make sure that it does *not* run as daemon."
exit 1
fi
kamailio="$kamailio -f kamailio.cfg -D -e -E"
;;
"freeswitch")
if [ -z "$(which freeswitch)" ]; then
echo "ERROR: freeswitch is not installed."
echo "Guide: https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster"
echo "After installing it, make sure that it does *not* run as daemon."
exit 1
fi
;;
"none")
;;
*)
echo "ERROR: unknown value "${SIPCON_SERVER}" for SIPCON_SERVER!"
exit 1
;;
esac
fi
sudo tcpdump -i $dev -n -w current_log/$dev.single.pcap -U not port 22 &
@ -123,7 +141,10 @@ if [ "x${MSC_MNCC}" != "xinternal" ]; then
sleep .2
term "$sipcon" SIPCON &
sleep .2
term "$kamailio" KAMAILIO &
case "${SIPCON_SERVER}" in
"kamailio") term "$kamailio" KAMAILIO &;;
"freeswitch") term "./freeswitch/freeswitch.sh" FREESWITCH &;;
esac
fi
#ssh bts rm /tmp/bts.log /tmp/pcu.log
@ -150,7 +171,7 @@ killall osmo-ggsn
if [ "x${MSC_MNCC}" != "xinternal" ]; then
# 'killall' seems to work only with the shortened name
killall osmo-sip-connec
killall kamailio
killall "${SIPCON_SERVER}"
fi