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conf: add config files to module directories

Browse Source 
The idea here is that the config files in the module directories will
serve as the canonical documentation of all module configuration
options.
This commit is contained in:
Ken Rice 2012-03-16 03:58:01 +00:00 committed by Travis Cross
parent f65140661c
commit cac6c262bb
83 changed files with 3108 additions and 0 deletions
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5
src/mod/applications/mod_abstraction/conf/autoload_configs/abstraction.conf.xml Normal file
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<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>

11
src/mod/applications/mod_blacklist/conf/autoload_configs/blacklist.conf.xml Normal file
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<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>

38
src/mod/applications/mod_callcenter/conf/autoload_configs/callcenter.conf.xml Normal file
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<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>

33
src/mod/applications/mod_cidlookup/conf/autoload_configs/cidlookup.conf.xml Normal file
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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>

209
src/mod/applications/mod_conference/conf/autoload_configs/conference.conf.xml Normal file
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<!-- 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 ae ejected from the conference -->
<param name="kicked-sound" value="conference/conf-kicked.wav"/>
<!-- File to play when the conference is locked -->
<param name="locked-sound" value="conference/conf-locked.wav"/>
<!-- File to play when the conference is locked during the call-->
<param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
<!-- File to play when the conference is unlocked during the call-->
<param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/>
<!-- File to play to prompt for a pin -->
<param name="pin-sound" value="conference/conf-pin.wav"/>
<!-- File to play to when the pin is invalid -->
<param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
<!-- Conference pin -->
<!--<param name="pin" value="12345"/>-->
<!--<param name="moderator-pin" value="54321"/>-->
<!-- Max number of times the user can be prompted for PIN -->
<!--<param name="pin-retries" value="3"/>-->
<!-- Default Caller ID Name for outbound calls -->
<param name="caller-id-name" value="$${outbound_caller_name}"/>
<!-- Default Caller ID Number for outbound calls -->
<param name="caller-id-number" value="$${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="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
-->
<!-- IVR digit machine timeouts -->
<!-- How much to wait between DTMF digits to match caller-controls -->
<!-- <param name="ivr-dtmf-timeout" value="500"/> -->
<!-- How much to wait for the first DTMF, 0 forever -->
<!-- <param name="ivr-input-timeout" value="0" /> -->
<!-- Delay before a conference is asked to be terminated -->
<!-- <param name="endconf-grace-time" value="120" /> -->
</profile>
<profile name="wideband">
<param name="domain" value="$${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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src/mod/applications/mod_conference/conf/dialplan/conference_dialplan.xml Normal file
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<!--
start a dynamic conference with the settings of the "default" conference profile in conference.conf.xml
-->
<extension name="nb_conferences">
<condition field="destination_number" expression="^(30\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@default"/>
</condition>
</extension>
<extension name="wb_conferences">
<condition field="destination_number" expression="^(31\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@wideband"/>
</condition>
</extension>
<extension name="uwb_conferences">
<condition field="destination_number" expression="^(32\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@ultrawideband"/>
</condition>
</extension>
<!-- MONO 48kHz conferences -->
<extension name="cdquality_conferences">
<condition field="destination_number" expression="^(33\d{2})$">
<action application="answer"/>
<action application="conference" data="$1-${domain_name}@cdquality"/>
</condition>
</extension>

5
src/mod/applications/mod_db/conf/autoload_configs/db.conf.xml Normal file
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<configuration name="db.conf" description="LIMIT DB Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>

10
src/mod/applications/mod_distributor/conf/autoload_configs/distributor.conf.xml Normal file
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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>

8
src/mod/applications/mod_distributor/conf/dialplan/distributor.xml Normal file
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<?xml version="1.0" encoding="utf-8"?>
<include>
<extension name="test gateway distro">
<condition field="destination_number" expression="^(.*)$">
<action application="bridge" data="sofia/gateway/${distributor(test)}/$1"/>
</condition>
</extension>
</include>

28
src/mod/applications/mod_easyroute/conf/autoload_configs/easyroute.conf.xml Normal file
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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>

27
src/mod/applications/mod_easyroute/conf/dialplan/easyroute.xml Normal file
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<?xml version="1.0" encoding="utf-8"?>
<include>
<!-- This example will test the limit value for the DID number being checked -->
<!-- If over limit, send 503 -->
<extension name="easyroute">
<condition field="destination_number" expression="^1?(\d{10})$" break="on-true">
<action application="easyroute" data="$1"/>
<action application="limit" data="db easyroute ${easy_group} ${easy_limit} LE-$1"/>
<action application="bridge" data="${easy_dialstring}"/>
</condition>
</extension>
<extension name="easyroute_limit_exceeded">
<condition field="destination_number" expression="^LE-(\d{10})$" break="on-true">
<action application="respond" data="503 Buzz Off You Ran Out of Channels"/>
</condition>
</extension>
<!-- Simple example - just grab the dialstring and send the call -->
<extension name="easyroute">
<condition field="destination_number" expression="^1?(\d{10})$" break="on-true">
<action application="set" data="dialstring={easyroute($1 dialstring)"/>
<action application="bridge" data="${easy_dialstring}"/>
</condition>
</extension>
<include>

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src/mod/applications/mod_enum/conf/autoload_configs/enum.conf.xml Normal file
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<configuration name="enum.conf" description="ENUM Module">
<settings>
<param name="default-root" value="e164.org"/>
<param name="default-isn-root" value="freenum.org"/>
<param name="query-timeout" value="10"/>
<param name="auto-reload" value="true"/>
</settings>
<routes>
<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/${use_profile}/$1"/>
<!--<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>-->
</routes>
</configuration>

8
src/mod/applications/mod_enum/conf/dialplan/99999_enum.xml Normal file
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<include>
<extension name="enum">
<condition field="${module_exists(mod_enum)}" expression="true"/>
<condition field="destination_number" expression="^(.*)$">
<action application="transfer" data="$1 enum"/>
</condition>
</extension>
</include>

10
src/mod/applications/mod_fifo/conf/autoload_configs/fifo.conf.xml Normal file
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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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src/mod/applications/mod_fifo/conf/dialplan/fifo.xml Normal file
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<!--
Parking extensions... transferring calls to 5900 will park them in a queue.
-->
<extension name="park">
<condition field="destination_number" expression="^5900$">
<action application="set" data="fifo_music=$${hold_music}"/>
<action application="fifo" data="5900@${domain_name} in"/>
</condition>
</extension>
<!--
Parking pickup extension. Calling 5901 will pickup the call.
-->
<extension name="unpark">
<condition field="destination_number" expression="^5901$">
<action application="answer"/>
<action application="fifo" data="5900@${domain_name} out nowait"/>
</condition>
</extension>

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src/mod/applications/mod_hash/conf/autoload_configs/hash.conf.xml Normal file
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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>

7
src/mod/applications/mod_hash/conf/dialplan/hash.xml Normal file
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<extension name="hash_demo">
<condition field="destination_number" expression="^hash_demo$">
<action application="hash" data="insert/realm/key/value"/>
<action application="hash" data="delete/realm/key"/>
<action application="set" data="var=${hash(select/realm/key)}"/>
</condition>
</extension>

0
src/mod/applications/mod_httapi/httapi.conf.xml → src/mod/applications/mod_httapi/conf/autoload_configs/httapi.conf.xml
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0
src/mod/applications/mod_httapi/mod_httapi_doc.txt → src/mod/applications/mod_httapi/docs/mod_httapi_doc.txt
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8
src/mod/applications/mod_http_cache/conf/autoload_configs/http_cache.conf.xml Normal file
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<configuration name="http_cache.conf" description="HTTP GET cache">
<settings>
<param name="max-urls" value="10000"/>
<param name="location" value="$${base_dir}/http_cache"/>
<param name="default-max-age" value="86400"/>
</settings>
</configuration>

7
src/mod/applications/mod_http_cache/conf/dialplan/http_cache.xml Normal file
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<include>
<extension "http_cache_demo">
<condition field="destination_number" expression="^(.*)$">
<action application="playback" data="${http_get(http://myserver.yo/media/hello_world.wav)}"/>
</condition>
</extension>
</include>

77
src/mod/applications/mod_ladspa/conf/dialplan/00_ladspa.xml Normal file
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<include>
<X-PRE-PROCESS cmd="set" data="AT_EPENT1=0 0 0 -1 -1 0 -1 0 -1 -1 0 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_EPENT2=1 1 1 -1 -1 1 -1 1 -1 -1 1 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_CPENT1=0 -1 -1 0 -1 0 0 0 -1 -1 0 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_CPENT2=1 -1 -1 1 -1 1 1 1 -1 -1 1 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_CMAJ1=0 -1 0 0 -1 0 -1 0 0 -1 0 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_CMAJ2=1 -1 1 1 -1 1 -1 1 1 -1 1 -1"/>
<X-PRE-PROCESS cmd="set" data="AT_BBLUES=1 -1 1 -1 -1 1 -1 1 1 1 -1 -1"/>
<X-PRE-PROCESS cmd="set" data="ATGPENT2=-1 1 -1 1 -1 1 -1 -1 1 -1 1 -1"/>
<extension name="101">
<condition field="destination_number" expression="^101$">
<!-- AUTOTALENT DEFAULTS -->
<!--
<action application="set" data="AT_TUNE=440"/>
<action application="set" data="AT_FIXED=0"/>
<action application="set" data="AT_PULL=0"/>
<action application="set" data="AT_A=0"/>
<action application="set" data="AT_Bb=-1"/>
<action application="set" data="AT_B=0"/>
<action application="set" data="AT_C=0"/>
<action application="set" data="AT_Db=-1"/>
<action application="set" data="AT_D=0"/>
<action application="set" data="AT_Eb=-1"/>
<action application="set" data="AT_E=0"/>
<action application="set" data="AT_F=0"/>
<action application="set" data="AT_Gb=-1"/>
<action application="set" data="AT_G=0"/>
<action application="set" data="AT_Ab=-1"/>
<action application="set" data="AT_AMOUNT=1"/>
<action application="set" data="AT_SMOOTH=0"/>
<action application="set" data="AT_SHIFT=0"/>
<action application="set" data="AT_OUTSCALE=0"/>
<action application="set" data="AT_LFODEPTH=0"/>
<action application="set" data="AT_LFORATE=5"/>
<action application="set" data="AT_LFOSHAPE=0"/>
<action application="set" data="AT_LFOSYMM=0"/>
<action application="set" data="AT_LFOQUANT=0"/>
<action application="set" data="AT_FCORR=0"/>
<action application="set" data="AT_FWARP=0"/>
<action application="set" data="AT_MIX=1"/>
-->
<action application="set" data="AT_TUNE=440"/>
<action application="set" data="AT_FIXED=0"/>
<action application="set" data="AT_PULL=0"/>
<action application="set" data="AT_AMOUNT=1"/>
<action application="set" data="AT_SMOOTH=0"/>
<action application="set" data="AT_SHIFT=1"/>
<action application="set" data="AT_OUTSCALE=0"/>
<action application="set" data="AT_LFODEPTH=0"/>
<action application="set" data="AT_LFORATE=5"/>
<action application="set" data="AT_LFOSHAPE=0"/>
<action application="set" data="AT_LFOSYMM=0"/>
<action application="set" data="AT_LFOQUANT=0"/>
<action application="set" data="AT_FCORR=0"/>
<action application="set" data="AT_FWARP=0"/>
<action application="set" data="AT_MIX=1"/>
<!-- you have to download the ladspa package and the desired plugins from their desired site -->
<action application="set"><![CDATA[ladspa_params=${AT_TUNE} ${AT_FIXED} ${AT_PULL} ${AT_EPENT2} ${AT_AMOUNT} ${AT_SMOOTH} ${AT_SHIFT} ${AT_OUTSCALE} ${AT_LFODEPTH} ${AT_LFORATE} ${AT_LFOSHAPE} ${AT_LFOSYMM} ${AT_LFOQUANT} ${AT_FCORR} ${AT_FWARP} ${AT_MIX}]]></action>
<action application="ladspa_run" data="r|autotalent||${ladspa_params}"/>
<action application="ladspa_run" data="r|tap_chorusflanger||"/>
<action application="ladspa_run" data="r|phasers_1217.so|autoPhaser|"/>
<action application="bridge" data="sofia/internal/888@conference.freeswitch.org"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_ladspa/docs/README Normal file
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mod_ladspa is a module that allow to use Linux Audio Developer's Simple Plugin API inside freeswitch in realtime.
see http://wiki.freeswitch.org/wiki/Mod_ladspa for installation and configuration documentation.
Please Note Currently this module only works on Linux.

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src/mod/applications/mod_lcr/conf/autoload_configs/lcr.conf.xml Normal file
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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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src/mod/applications/mod_lcr/conf/dialplan/lcr.xml Normal file
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<include>
<extension name="Outbound to PSTN 11 Digits">
<condition field="destination_number" expression="^(1[2-9][0-9]{2}[2-9]{7})$">
<action application="lcr" data="$1"/>
<action application="bridge" data="${lcr_auto_route}"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_limit/conf/autoload_configs/limit.conf.xml Normal file
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<configuration name="limit.conf" description="LIMIT Configuration">
<settings>
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
</settings>
</configuration>

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src/mod/applications/mod_limit/conf/dialplan/limit.xml Normal file
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<include>
<extension name="limit_exceeded">
<condition field="destination_number" expression="^limit_exceeded$">
<action application="playback" data="/sounds/overthelimit.wav"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="limit" continue="true">
<condition>
<action application="limit" data="db $${domain} ${sip_auth_username} ${max_calls}"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_memcache/conf/autoload_configs/memcache.conf.xml Normal file
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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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src/mod/applications/mod_memcache/conf/dialplan/memcache.xml Normal file
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<include>
<extension name="memcache_examples">
<condition field="destination_number" expression="^memcache$">
<action application="set" data="foo=${memcache(get key)}"/>
<action application="set" data="ignore=${memcache(set key value)}"/>
<action application="set" data="ignore=${memcache(set key ${channel_var})}"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_mongo/conf/autoload_configs/mongo.conf.xml Normal file
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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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src/mod/applications/mod_nibblebill/conf/autoload_configs/nibblebill.conf.xml Normal file
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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="db_username" value="bandwidth.com"/>
<param name="db_password" value="password"/>
<param name="db_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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src/mod/applications/mod_nibblebill/conf/dialplan/nibblebill.xml Normal file
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<include>
<extension name="tollfree800">
<condition field="destination_number" expression="^1{0,1}(800\d{7})$">
<action application="set" data="nibble_account=${accountcode}"/>
<action application="set" data="nibble_rate=0"/>
<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
</condition>
</extension>
<extension name="special919rate">
<condition field="destination_number" expression="^1{0,1}(919\d{7})$">
<action application="set" data="nibble_account=${accountcode}"/>
<action application="set" data="nibble_rate=0.07"/>
<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
</condition>
</extension>
<extension name="domestic">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<action application="set" data="nibble_account=${accountcode}"/>
<action application="set" data="nibble_rate=0.05"/>
<action application="bridge" data="sofia/gateway/bandwidth.com/$1"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_osp/conf/osp.conf.xml Normal file
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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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src/mod/applications/mod_redis/conf/autoload_configs/redis.conf.xml Normal file
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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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src/mod/applications/mod_rss/conf/autoload_configs/rss.conf.xml Normal file
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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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src/mod/applications/mod_rss/conf/dialplan/rss.xml Normal file
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<include>
<extension name="rss">
<condition field="destination_number" expression="^rss$">
<action application="rss"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_rss/script/news.js Normal file
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if (session.ready()) {
session.answer();
session.speak("cepstral","David","Please wait while we refresh the RSS feeds.")
fetchURLFile("http://weather.yahooapis.com/forecastrss?p=60610","rss/weather.rss");
fetchURLFile("http://rss.news.yahoo.com/rss/topstories","rss/yahootop.rss");
fetchURLFile("http://rss.news.yahoo.com/rss/science","rss/yahoosci.rss");
fetchURLFile("http://rss.news.yahoo.com/rss/business","rss/yahoobus.rss");
fetchURLFile("http://rss.news.yahoo.com/rss/entertainment","rss/yahooent.rss");
fetchURLFile("http://rss.slashdot.org/Slashdot/slashdot","rss/slashdot.rss");
fetchURLFile("http://www.freeswitch.org/xml.php","rss/freeswitch.rss");
}

16
src/mod/applications/mod_snom/conf/dialplan/snom_demo.xml Normal file
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<include>
<extension name="snom-demo-1">
<condition field="destination_number" expression="^9000$">
<!-- <key> <light> <label> <user> <host> <profile> <action_name> <action> -->
<action application="eval" data="${snom_bind_key(2 on DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message api+uuid_transfer ${uuid} 9001)}"/>
<action application="playback" data="$${hold_music}"/>
</condition>
</extension>
<extension name="snom-demo-2">
<condition field="destination_number" expression="^9001$">
<action application="eval" data="${snom_bind_key(2 off DND ${sip_from_user} ${sip_from_host} ${sofia_profile_name} message notused)}"/>
<action application="transfer" data="3000"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_spandsp/conf/autoload_configs/spandsp.conf.xml Normal file
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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"/>
<!-- 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="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="480" freq2="620" min="224" max="272"/>
<element freq1="0" freq2="0" min="224" max="272"/>
</tone>
<tone name="BUSY_TONE">
<element freq1="480" freq2="620" min="464" max="516"/>
<element freq1="0" freq2="0" min="464" max="516"/>
</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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src/mod/applications/mod_spandsp/conf/config.FS0 Normal file
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CountryCode: 1
AreaCode: 800
FAXNumber: +1.800.555.1212
LongDistancePrefix: 1
InternationalPrefix: 011
DialStringRules: etc/dialrules
ServerTracing: 0xFFF
SessionTracing: 0xFFF
RecvFileMode: 0600
LogFileMode: 0600
DeviceMode: 0600
RingsBeforeAnswer: 1
SpeakerVolume: off
GettyArgs: "-h %l dx_%s"
LocalIdentifier: "FS"
TagLineFont: etc/lutRS18.pcf
TagLineFormat: "From %%l|%c|Page %%P of %%T"
MaxRecvPages: 200
#
#
# Modem-related stuff: should reflect modem command interface
# and hardware connection/cabling (e.g. flow control).
#
ModemType: Class1 # use this to supply a hint
#
# Enabling this will use the hfaxd-protocol to set Caller*ID
#
#ModemSetOriginCmd: AT+VSID="%s","%d"
#
# If "glare" during initialization becomes a problem then take
# the modem off-hook during initialization, and then place it
# back on-hook when done.
#
#ModemResetCmds: "ATH1\nAT+VCID=1" # enables CallID display
#ModemReadyCmds: ATH0
Class1AdaptRecvCmd: AT+FAR=1
Class1TMConnectDelay: 400 # counteract quick CONNECT response
#
# If you have trouble with V.17 receiving or sending,
# you may want to enable one of these, respectively.
#
#Class1RMQueryCmd: "!24,48,72,96" # enable this to disable V.17 receiving
#Class1TMQueryCmd: "!24,48,72,96" # enable this to disable V.17 sending
#
# You'll likely want Caller*ID display (also displays DID) enabled.
#
ModemResetCmds: AT+VCID=1 # enables CallID display
#
# The pty does not support changing parity.
#
PagerTTYParity: none
#
# If you are "missing" Caller*ID data on some calls (but not all)
# and if you do not have adequate glare protection you may want to
# not answer based on RINGs, but rather enable the CallIDAnswerLength
# for NDID, disable AT+VCID=1 and do this:
#
#RingsBeforeAnswer: 0
#ModemRingResponse: AT+VRID=1
# Uncomment DATE and TIME if you really want them, but you probably don't.
#CallIDPattern: "DATE="
#CallIDPattern: "TIME="
CallIDPattern: "NMBR="
CallIDPattern: "NAME="
CallIDPattern: "ANID="
#CallIDPattern: "USER=" # username provided by call
#CallIDPattern: "PASS=" # password provided by call
#CallIDPattern: "CDID=" # DID context in call
CallIDPattern: "NDID="
#CallIDAnswerLength: 4

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src/mod/applications/mod_spandsp/conf/dialplan/spandsp.xml Normal file
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<extension name="fax_receive">
<condition field="destination_number" expression="^9178$">
<action application="answer" />
<action application="playback" data="silence_stream://2000"/>
<action application="rxfax" data="/tmp/rxfax.tif"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="fax_transmit">
<condition field="destination_number" expression="^9179$">
<action application="txfax" data="/tmp/txfax.tif"/>
<action application="hangup"/>
</condition>
</extension>
<extension name="fax_receive">
<condition field="destination_number" expression="^9978$">
<action application="answer" />
<action application="playback" data="silence_stream://2000"/>
<action application="set" data="fax_enable_t38_request=true"/>
<action application="set" data="fax_enable_t38=true"/>
<action application="rxfax" data="/tmp/FAX-${uuid}.tif"/>
<action application="hangup"/>
</condition>
</extension>

24
src/mod/applications/mod_valet_parking/conf/dialplan/valet_parking.xml Normal file
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<include>
<!--
Valet park retrieval, works with valet_park extension below.
Retrieve a valet parked call by dialing 6000 + park number + #
-->
<extension name="valet_park">
<condition field="destination_number" expression="^(6000)$">
<action application="answer"/>
<action application="valet_park" data="valet_parking_lot ask 1 11 10000 ivr/ivr-enter_ext_pound.wav"/>
</condition>
</extension>
<!--
Valet park 6001-6099. Blind x-fer to 6001, 6002, etc. to valet park the call.
Dial 6001, 6002, etc. to retrieve a call that is already valet parked.
After call is retrieved, park extension is free for another call.
-->
<extension name="valet_park">
<condition field="destination_number" expression="^(60\d[1-9])$">
<action application="answer"/>
<action application="valet_park" data="valet_parking_lot $1"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_vmd/conf/dialplan/vmd.xml Normal file
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<include>
<extension name="vmdtest">
<condition field="destination_number" expression="^vmd(\d{10})$">
<action application="answer"/>
<action application="info"/>
<action application="vmd"/>
<action application="sleep" data="25000"/>
<action application="info"/> <!-- Look for chan var "vmd_detect" here -->
<action application="hangup"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_vmd/scripts/vmd.js Normal file
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function onInput(s, type, obj, arg)
{
try
{
if(type == "dtmf")
{
console_log("info", "DTMF digit: "+s.name+" ["+obj.digit+"] len ["+obj.duration+"]\n");
}
else if(type == "event" && session.getVariable("vmd_detect") == "TRUE")
{
console_log("info", "Voicemail Detected\n");
}
}
catch(e)
{
console_log("err", e + "\n");
}
return true;
}
session.answer();
session.execute("vmd", "start");
while(session.ready())
{
session.streamFile(argv[0], onInput);
}

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src/mod/applications/mod_vmd/scripts/vmd.lua Normal file
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local human_detected = false;
local voicemail_detected = false;
function onInput(session, type, obj)
if type == "dtmf" and obj['digit'] == '1' and human_detected == false then
human_detected = true;
return "break";
end
if type == "event" and voicemail_detected == false then
voicemail_detected = true;
return "break";
end
end
session:setInputCallback("onInput");
session:execute("vmd","start");

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src/mod/applications/mod_voicemail/conf/autoload_configs/voicemail.conf.xml Normal file
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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"/>
<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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src/mod/applications/mod_voicemail/conf/dialplan/voicemail.xml Normal file
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<include>
<!-- This is a _VERY_ abbreviated demo for voicemail. A better demo would be useful -->
<!-- voicemail main extension -->
<extension name="vmain">
<condition field="destination_number" expression="^vmain$|^4000$|^\*98$">
<action application="answer"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="check default ${domain_name}"/>
</condition>
</extension>
<extension name="Local_Voicemail">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="export" data="dialed_extension=$1"/>
<action application="export" data="domain=default"/>
<action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/>
</condition>
</extension>
</include>

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src/mod/applications/mod_voicemail/conf/notify-voicemail.tpl Normal file
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From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}@${voicemail_domain}>
Date: ${RFC2822_DATE}
To: <${voicemail_notify_email}>
Subject: Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}
X-Priority: ${voicemail_priority}
X-Mailer: FreeSWITCH
Content-Type: multipart/alternative;
boundary="000XXX000"
--000XXX000
Content-Type: text/plain; charset=ISO-8859-1; Format=Flowed
Content-Disposition: inline
Content-Transfer-Encoding: 7bit
Created: ${voicemail_time}
From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}>
Duration: ${voicemail_message_len}
Account: ${voicemail_account}@${voicemail_domain}
--000XXX000
Content-Type: text/html; charset=ISO-8859-1
Content-Disposition: inline
Content-Transfer-Encoding: 7bit
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<META http-equiv="Content-Type" content="text/html; charset=ISO-8859-1">
<title>Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}</title>
<meta content="text/html; charset=iso-8859-1" http-equiv="content-type"/>
</head>
<body>
<font face=arial>
<b>Message From "${voicemail_caller_id_name}" <A HREF="tel:${voicemail_caller_id_number}">${voicemail_caller_id_number}</A></b><br>
<hr noshade size=1>
Created: ${voicemail_time}<br>
Duration: ${voicemail_message_len}<br>
Account: ${voicemail_account}@${voicemail_domain}<br>
</font>
</body>
</html>
--000XXX000--

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src/mod/applications/mod_voicemail/conf/voicemail.tpl Normal file
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From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}@${voicemail_domain}>
Date: ${RFC2822_DATE}
To: <${voicemail_email}>
Subject: Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}
X-Priority: ${voicemail_priority}
X-Mailer: FreeSWITCH
Content-Type: multipart/alternative;
boundary="000XXX000"
--000XXX000
Content-Type: text/plain; charset=ISO-8859-1; Format=Flowed
Content-Disposition: inline
Content-Transfer-Encoding: 7bit
Created: ${voicemail_time}
From: "${voicemail_caller_id_name}" <${voicemail_caller_id_number}>
Duration: ${voicemail_message_len}
Account: ${voicemail_account}@${voicemail_domain}
--000XXX000
Content-Type: text/html; charset=ISO-8859-1
Content-Disposition: inline
Content-Transfer-Encoding: 7bit
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<META http-equiv="Content-Type" content="text/html; charset=ISO-8859-1">
<title>Voicemail from "${voicemail_caller_id_name}" <${voicemail_caller_id_number}> ${voicemail_message_len}</title>
<meta content="text/html; charset=iso-8859-1" http-equiv="content-type"/>
</head>
<body>
<font face=arial>
<b>Message From "${voicemail_caller_id_name}" <A HREF="tel:${voicemail_caller_id_number}">${voicemail_caller_id_number}</A></b><br>
<hr noshade size=1>
Created: ${voicemail_time}<br>
Duration: ${voicemail_message_len}<br>
Account: ${voicemail_account}@${voicemail_domain}<br>
</font>
</body>
</html>
--000XXX000--

208
src/mod/applications/mod_voicemail_ivr/conf/autoload_configs/voicemail_ivr.conf.xml Normal file
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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_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">
<phrases>
<phrase name="msg_count" value="message_count@mtvoicemail" />
<phrase name="say_date" value="say_date_event@mtvoicemail" />
<phrase name="say_msg_number" value="say_message_number@mtvoicemail" />
<phrase name="menu_options" value="menu@mtvoicemail" />
</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="3" action="save_msg" variable="VM-Key-Main-Save-File" />
<key dtmf="2" action="callback" variable="VM-Key-Main-Callback" />
<key dtmf="5" action="menu:std_preference" />
<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="#" 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="3" 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="3" 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="3" 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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src/mod/asr_tts/mod_cepstral/conf/cepstral.conf.xml Normal file
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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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src/mod/dialplans/mod_dialplan_asterisk/conf/extensions.conf Normal file
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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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src/mod/endpoints/mod_alsa/conf/autoload_configs/alsa.conf.xml Normal file
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<configuration name="alsa.conf" description="Soundcard Endpoint">
<settings>
<!--Default dialplan and caller-id info -->
<param name="dialplan" value="XML"/>
<param name="cid-name" value="N800 Alsa"/>
<param name="cid-num" value="5555551212"/>
<!--audio sample rate and interval -->
<param name="sample-rate" value="8000"/>
<param name="codec-ms" value="20"/>
</settings>
</configuration>

218
src/mod/endpoints/mod_portaudio/conf/autoload_configs/portaudio.conf.xml Normal file
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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"/>
</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="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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src/mod/endpoints/mod_skinny/conf/autoload_configs/skinny.conf.xml Normal file
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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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src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns.xml Normal file
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<?xml version="1.0" encoding="utf-8"?>
<!--
NOTICE:
This context is used for skinny to match dialed number
The special applications:
- skinny-process tells skinny to process the call (route, set call forwarding, ...)
- skinny-drop tells skinny to drop the call
- skinny-wait tells skinny to wait 'data' seconds for more numbers before drop
-->
<!-- http://wiki.freeswitch.org/wiki/Mod_skinny -->
<include>
<context name="skinny-patterns">
<!--
Wait 10 seconds for another digit by default
-->
<extension name="Default">
<condition>
<action application="skinny-wait" data="10"/>
</condition>
</extension>
<!--
You can place files in the skinny-patterns directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="skinny-patterns/*.xml"/>
</context>
</include>

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src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/20-Demo.xml Normal file
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<include>
<extension name="Demo">
<condition field="destination_number" expression="^(9[0-9]{3})$">
<action application="skinny-process"/>
</condition>
</extension>
</include>

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src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/20-Local_extension.xml Normal file
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<include>
<extension name="Local_Extension">
<condition field="destination_number" expression="^(10[01][0-9])$">
<action application="skinny-process"/>
</condition>
</extension>
<extension name="Local_Extension_Skinny">
<condition field="destination_number" expression="^(11[01][0-9])$">
<action application="skinny-process"/>
</condition>
</extension>
</include>

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src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/90-External.xml Normal file
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<include>
<extension name="External">
<condition field="destination_number" expression="^(0[0-9]{10,})$">
<action application="skinny-process"/>
</condition>
</extension>
</include>

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src/mod/endpoints/mod_skinny/conf/dialplan/skinny-patterns/99-Default_Drop.xml Normal file
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<include>
<!-- drop long dialed number -->
<extension name="Default_Drop">
<condition field="destination_number" expression="^(.{14,})$">
<action application="skinny-drop"/>
</condition>
</extension>
</include>

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src/mod/endpoints/mod_skinny/conf/directory/default/skinny-example.xml Normal file
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<include>
<user id="SEP001120AABBCC">
<params>
<!-- for devices requesting firmware via SCCP, like ATA186
<param name="skinny-firmware-version" value="ATA030101SCCP04"
<param name="skinny-soft-key-set-set" value="default"
-->
<param name="foo" value="bar"/>
</params>
<skinny>
<buttons>
<!--
position: 1..
type: one of line, speed-dial
label: button label
-->
<!--
value is the directory number (or user)
caller-name is shown to the calling party during call
-->
<button position="1" type="Line" label="Line 1" value="1101" caller-name="Calling as 1101"/>
<button position="3" type="Line" label="Shared Line 10" value="1110" caller-name="Calling as 1110"/>
<!--
value is the directory number to call
-->
<button position="5" type="SpeedDial" label="Call 1001" value="1001"/>
<!--
value is the URL
-->
<button position="6" type="ServiceUrl" label="Some URL" value="http://phone-xml.berbee.com/menu.xml"/>
</buttons>
</skinny>
</user>
</include>

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<profile name="internal">
<settings>
<param name="domain" value="$${domain}"/>
<param name="ip" value="$${local_ip_v4}"/>
<param name="port" value="2000"/>
<param name="patterns-dialplan" value="XML"/>
<param name="patterns-context" value="skinny-patterns"/>
<param name="dialplan" value="XML"/>
<param name="context" value="default"/>
<param name="keep-alive" value="60"/>
<param name="date-format" value="D/M/Y"/>
<param name="odbc-dsn" value=""/>
<param name="debug" value="4"/>
<param name="auto-restart" value="true"/>
</settings>
<soft-key-set-sets>
<soft-key-set-set name="default">
<soft-key-set name="KeySetOnHook" value="SoftkeyNewcall,SoftkeyRedial"/>
<soft-key-set name="KeySetConnected" value="SoftkeyEndcall,SoftkeyHold,SoftkeyNewcall,SoftkeyTransfer"/>
<soft-key-set name="KeySetOnHold" value="SoftkeyNewcall,SoftkeyResume,SoftkeyEndcall"/>
<soft-key-set name="KeySetRingIn" value="SoftkeyAnswer,SoftkeyEndcall,SoftkeyNewcall"/>
<soft-key-set name="KeySetOffHook" value=",SoftkeyRedial,SoftkeyEndcall"/>
<soft-key-set name="KeySetConnectedWithTransfer" value="SoftkeyEndcall,SoftkeyHold,SoftkeyNewcall,SoftkeyTransfer"/>
<soft-key-set name="KeySetDigitsAfterDialingFirstDigit" value="SoftkeyBackspace,,SoftkeyEndcall"/>
<!-- <soft-key-set name="KeySetConnectedWithConference" value=""/> -->
<soft-key-set name="KeySetRingOut" value=",,SoftkeyEndcall,SoftkeyTransfer"/>
<soft-key-set name="KeySetOffHookWithFeatures" value=",SoftkeyRedial,SoftkeyEndcall"/>
<soft-key-set name="KeySetInUseHint" value="SoftkeyNewcall,SoftkeyRedial"/>
</soft-key-set-set>
</soft-key-set-sets>
<device-types>
<device-type id="Cisco ATA 186">
<param name="firmware-version" value="ATA030101SCCP04"/>
</device-type>
</device-types>
</profile>

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src/mod/endpoints/mod_sofia/conf/sip_profiles/external.xml Normal file
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<profile name="external">
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!-- This profile is only for outbound registrations to providers -->
<gateways>
<X-PRE-PROCESS cmd="include" data="external/*.xml"/>
</gateways>
<aliases>
<!--
<alias name="outbound"/>
<alias name="nat"/>
-->
</aliases>
<domains>
<domain name="all" alias="false" parse="true"/>
</domains>
<settings>
<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"/>
<param name="rfc2833-pt" value="101"/>
<!-- RFC 5626 : Send reg-id and sip.instance -->
<!--<param name="enable-rfc-5626" value="true"/> -->
<param name="sip-port" value="$${external_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="rtp-timer-name" value="soft"/>
<!--<param name="enable-100rel" value="true"/>-->
<!--<param name="disable-srv503" value="true"/>-->
<!-- This could be set to "passive" -->
<param name="local-network-acl" value="localnet.auto"/>
<param name="manage-presence" value="false"/>
<!-- used to share presence info across sofia profiles
manage-presence needs to be set to passive on this profile
if you want it to behave as if it were the internal profile
for presence.
-->
<!-- Name of the db to use for this profile -->
<!--<param name="dbname" value="share_presence"/>-->
<!--<param name="presence-hosts" value="$${domain}"/>-->
<!--<param name="force-register-domain" value="$${domain}"/>-->
<!--all inbound reg will stored in the db using this domain -->
<!--<param name="force-register-db-domain" value="$${domain}"/>-->
<!-- ************************************************* -->
<!--<param name="aggressive-nat-detection" value="true"/>-->
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<!--
DO NOT USE HOSTNAMES, ONLY IP ADDRESSES IN THESE SETTINGS!
-->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!--<param name="enable-3pcc" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${external_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. (5081 will be used if unspecified) -->
<param name="tls-sip-port" value="$${external_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${external_ssl_dir}"/>
<!-- 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}"/>
</settings>
</profile>

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src/mod/endpoints/mod_sofia/conf/sip_profiles/external/example.xml vendored Normal file
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<include>
<!--<gateway name="asterlink.com">-->
<!--/// account username *required* ///-->
<!--<param name="username" value="cluecon"/>-->
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<!--<param name="realm" value="asterlink.com"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--<param name="from-user" value="cluecon"/>-->
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--<param name="from-domain" value="asterlink.com"/>-->
<!--/// account password *required* ///-->
<!--<param name="password" value="2007"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<!--<param name="proxy" value="asterlink.com"/>-->
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<!--<param name="register" value="false"/>-->
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
<!--</gateway>-->
<!--rfc5626 : Abilitazione rfc5626 ///-->
<!--<param name="rfc-5626" value="true"/>-->
<!--rfc5626 : extra sip params to send in the contact-->
<!--<param name="reg-id" value="1"/>-->
</include>

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<profile name="internal-ipv6">
<!--
This is an example of a sofia profile setup to listen on IPv6.
-->
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<!--aliases are other names that will work as a valid profile name for this profile-->
<settings>
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="$${internal_sip_port}"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<!-- ip address to use for rtp -->
<param name="rtp-ip" value="$${local_ip_v6}"/>
<!-- ip address to bind to -->
<param name="sip-ip" value="$${local_ip_v6}"/>
<param name="hold-music" value="$${hold_music}"/>
<!--<param name="enable-100rel" value="false"/>-->
<!--<param name="disable-srv503" value="true"/>-->
<param name="apply-inbound-acl" value="domains"/>
<!--<param name="apply-register-acl" value="domains"/>-->
<!--<param name="dtmf-type" value="info"/>-->
<param name="record-template" value="$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<!--enable to use presence and mwi -->
<param name="manage-presence" value="true"/>
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<!--<param name="max-proceeding" value="1000"/>-->
<!--session timers for all call to expire after the specified seconds -->
<!--<param name="session-timeout" value="1800"/>-->
<!--<param name="multiple-registrations" value="true"/>-->
<!--set to 'greedy' if you want your codec list to take precedence -->
<param name="inbound-codec-negotiation" value="generous"/>
<!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--<param name="unregister-on-options-fail" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${internal_ssl_enable}"/>
<!-- 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="$${internal_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="$${sip_tls_version}"/>
<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
<!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
<!-- <param name="accept-blind-reg" value="true"/> -->
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<!-- <param name="accept-blind-auth" value="true"/> -->
<!-- suppress CNG on this profile or per call with the 'suppress_cng' variable -->
<!-- <param name="suppress-cng" value="true"/> -->
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
<param name="auth-calls" value="$${internal_auth_calls}"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
<!-- 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="$${domain}"/>
<!--all inbound reg will stored in the db using this domain -->
<param name="force-register-db-domain" value="$${domain}"/>
<!-- disable register and transfer which may be undesirable in a public switch -->
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!--<param name="stun-enabled" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
<!--<param name="stun-auto-disable" value="true"/>-->
<!-- the following can be used as workaround with bogus SRV/NAPTR records -->
<!--<param name="disable-srv" value="false" />-->
<!--<param name="disable-naptr" value="false" />-->
</settings>
</profile>

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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 -->
<gateways>
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
</gateways>
<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>
<!--
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="$${internal_sip_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="$${local_ip_v4}"/>
<!-- ip address to bind to, DO NOT USE HOSTNAMES ONLY IP ADDRESSES -->
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="apply-nat-acl" value="nat.auto"/>
<!-- (default true) set to false if you do not wish to have called party info in 1XX responses -->
<!-- <param name="cid-in-1xx" value="false"/> -->
<!-- extended info parsing -->
<!-- <param name="extended-info-parsing" value="true"/> -->
<!--<param name="aggressive-nat-detection" value="true"/>-->
<!--
There are known issues (asserts and segfaults) when 100rel is enabled.
It is not recommended to enable 100rel at this time.
-->
<!--<param name="enable-100rel" value="true"/>-->
<!-- uncomment if you don't wish to try a next SRV destination on 503 response -->
<!-- RFC3263 Section 4.3 -->
<!--<param name="disable-srv503" value="true"/>-->
<!-- Enable Compact SIP headers. -->
<!--<param name="enable-compact-headers" value="true"/>-->
<!--
enable/disable session timers
-->
<!--<param name="enable-timer" value="false"/>-->
<!--<param name="minimum-session-expires" value="120"/>-->
<param name="apply-inbound-acl" value="domains"/>
<!--
This defines your local network, by default we detect your local network
and create this localnet.auto ACL for this.
-->
<param name="local-network-acl" value="localnet.auto"/>
<!--<param name="apply-register-acl" value="domains"/>-->
<!--<param name="dtmf-type" value="info"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-message-query-on-register" value="true"/>-->
<!-- 'true' means every time 'first-only' means on the first register -->
<!--<param name="send-presence-on-register" value="first-only"/> -->
<!-- Caller-ID type (choose one, can be overridden by inbound call type and/or sip_cid_type channel variable -->
<!-- Remote-Party-ID header -->
<!--<param name="caller-id-type" value="rpid"/>-->
<!-- P-*-Identity family of headers -->
<!--<param name="caller-id-type" value="pid"/>-->
<!-- neither one -->
<!--<param name="caller-id-type" value="none"/>-->
<param name="record-path" value="$${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="true"/>
<!-- 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="$${presence_privacy}"/>
<!-- ************************************************* -->
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<!--<param name="max-proceeding" value="1000"/>-->
<!--session timers for all call to expire after the specified seconds -->
<!--<param name="session-timeout" value="1800"/>-->
<!-- Can be 'true' or 'contact' -->
<!--<param name="multiple-registrations" value="contact"/>-->
<!--set to 'greedy' if you want your codec list to take precedence -->
<param name="inbound-codec-negotiation" value="generous"/>
<!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--<param name="unregister-on-options-fail" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="$${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="$${internal_tls_port}"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
<!-- 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"/>-->
<!--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"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
<!--<param name="inbound-late-negotiation" 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="auto-nat"/>
<param name="ext-sip-ip" value="auto-nat"/>
<!-- 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}"/>
<!--<param name="delete-subs-on-register" value="false"/>-->
<!-- 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"/>-->
<!-- 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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src/mod/endpoints/mod_sofia/conf/sip_profiles/internal/example.xml Normal file
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<include>
<!--<gateway name="asterlink.com">-->
<!--/// account username *required* ///-->
<!--<param name="username" value="cluecon"/>-->
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<!--<param name="realm" value="asterlink.com"/>-->
<!--/// username to use in from: *optional* same as username, if blank ///-->
<!--<param name="from-user" value="cluecon"/>-->
<!--/// domain to use in from: *optional* same as realm, if blank ///-->
<!--<param name="from-domain" value="asterlink.com"/>-->
<!--/// account password *required* ///-->
<!--<param name="password" value="2007"/>-->
<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->
<!--/// proxy host: *optional* same as realm, if blank ///-->
<!--<param name="proxy" value="asterlink.com"/>-->
<!--/// send register to this proxy: *optional* same as proxy, if blank ///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<!--<param name="register" value="false"/>-->
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound calls via this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!-- Put the extension in the contact -->
<!--<param name="extension-in-contact" value="true"/>-->
<!--send an options ping every x seconds, failure will unregister and/or mark it down-->
<!--<param name="ping" value="25"/>-->
<!--<param name="cid-type" value="rpid"/>-->
<!--</gateway>-->
</include>

22
src/mod/event_handlers/mod_cdr_csv/conf/autoload_configs/cdr_csv.conf.xml Normal file
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<configuration name="cdr_csv.conf" description="CDR CSV Format">
<settings>
<!-- 'cdr-csv' will always be appended to log-base -->
<!--<param name="log-base" value="/var/log"/>-->
<param name="default-template" value="example"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
<param name="rotate-on-hup" value="true"/>
<!-- may be a b or ab -->
<param name="legs" value="a"/>
<!-- Only log in Master.csv -->
<!-- <param name="master-file-only" value="true"/> -->
</settings>
<templates>
<template name="sql">INSERT INTO cdr VALUES ("${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}");</template>
<template name="example">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}"</template>
<template name="snom">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}", "${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${call_clientcode}","${sip_rtp_rxstat}","${sip_rtp_txstat}","${sofia_record_file}"</template>
<template name="linksys">"${caller_id_name}","${caller_id_number}","${destination_number}","${context}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${uuid}","${bleg_uuid}","${accountcode}","${read_codec}","${write_codec}","${sip_user_agent}","${sip_p_rtp_stat}"</template>
<template name="asterisk">"${accountcode}","${caller_id_number}","${destination_number}","${context}","${caller_id}","${channel_name}","${bridge_channel}","${last_app}","${last_arg}","${start_stamp}","${answer_stamp}","${end_stamp}","${duration}","${billsec}","${hangup_cause}","${amaflags}","${uuid}","${userfield}"</template>
</templates>
</configuration>

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src/mod/event_handlers/mod_cdr_mongodb/conf/autoload_configs/cdr_mongodb.conf.xml Normal file
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<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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src/mod/event_handlers/mod_cdr_pg_csv/conf/autoload_configs/cdr_pg_csv.conf.xml Normal file
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<configuration name="cdr_pg_csv.conf" description="CDR PG CSV Format">
<settings>
<!-- See parameters for PQconnectdb() at http://www.postgresql.org/docs/8.4/static/libpq-connect.html -->
<param name="db-info" value="host=localhost dbname=cdr connect_timeout=10" />
<!-- CDR table name -->
<!--<param name="db-table" value="cdr"/>-->
<!-- Log a-leg (a), b-leg (b) or both (ab) -->
<param name="legs" value="a"/>
<!-- Directory in which to spool failed SQL inserts -->
<!-- <param name="spool-dir" value="$${base_dir}/log/cdr-pg-csv"/> -->
<!-- Disk spool format if DB connection/insert fails - csv (default) or sql -->
<param name="spool-format" value="csv"/>
<param name="rotate-on-hup" value="true"/>
<!-- This is like the info app but after the call is hung up -->
<!--<param name="debug" value="true"/>-->
</settings>
<schema>
<field var="local_ip_v4"/>
<field var="caller_id_name"/>
<field var="caller_id_number"/>
<field var="destination_number"/>
<field var="context"/>
<field var="start_stamp"/>
<field var="answer_stamp"/>
<field var="end_stamp"/>
<field var="duration" quote="false"/>
<field var="billsec" quote="false"/>
<field var="hangup_cause"/>
<field var="uuid"/>
<field var="bleg_uuid"/>
<field var="accountcode"/>
<field var="read_codec"/>
<field var="write_codec"/>
<!-- <field var="sip_hangup_disposition"/> -->
<!-- <field var="ani"/> -->
</schema>
</configuration>

0
src/mod/event_handlers/mod_cdr_pg_csv/create.sql → src/mod/event_handlers/mod_cdr_pg_csv/scripts/create.sql
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18
src/mod/event_handlers/mod_cdr_sqlite/conf/autoload_configs/cdr_sqlite.conf.xml Normal file
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<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>

23
src/mod/event_handlers/mod_erlang_event/conf/autoload_configs/erlang_event.conf.xml Normal file
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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>

14
src/mod/event_handlers/mod_event_multicast/conf/autoload_configs/event_multicast.conf.xml Normal file
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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>

9
src/mod/event_handlers/mod_event_socket/conf/autoload_configs/event_socket.conf.xml Normal file
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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>

53
src/mod/event_handlers/mod_json_cdr/conf/autoload_configs/json_cdr.conf.xml Normal file
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<include>
<configuration name="json_cdr.conf" description="JSON CDR">
<settings>
<!-- Global parameters -->
<param name="log-b-leg" value="true"/>
<param name="prefix-a-leg" value="false"/>
<!-- Whether to URL encode the individual JSON values. Defaults to true, set to false for standard JSON. -->
<param name="encode-values" value="true"/>
<!-- Normally if url and log-dir are present, url is attempted first and log-dir second.
This options allows to do both systematically. -->
<param name="log-http-and-disk" value="false"/>
<!-- File logging -->
<!-- Directory where to create the "json_cdr" directory used to store JSON CDRs. Leave empty for no file logging. -->
<!-- Might be overriden by a channel variable "json_cdr_base". -->
<param name="log-dir" value=""/>
<!-- Whether to rotate file CDRs. -->
<param name="rotate" value="false"/>
<!-- HTTP(S) logging -->
<!-- URL where to POST JSON CDRs. Leave empty for no URL logging. Up to 20 URLs may be specified. -->
<param name="url" value=""/>
<!-- Authentication scheme for the above URL. May be one of basic|digest|NTLM|GSS-NEGOTIATE|any-->
<param name="auth-scheme" value="basic"/>
<!-- Credentials in the form username:password if auth-scheme is used. Leave empty for no authentication. -->
<param name="cred" value="string"/>
<!-- Whether to base64 encode the entire JSON document before POSTing it. -->
<param name="encode" value="base64|true|false"/>
<!-- Number of retries in case of failure. Each specified URL is tried in turn. -->
<param name="retries" value="0"/>
<!-- Delay between retries (ms). -->
<param name="delay" value="5000"/>
<!-- Disable streaming if the server doesn't support it. -->
<param name="disable-100-continue" value="false"/>
<!-- If web posting failed, the CDR is written to a file. -->
<!-- Error log dir ("json_cdr" is appended). Up to 20 may be specified. Default to log-dir if none is specified. -->
<param name="err-log-dir" value=""/>
<!-- SSL options -->
<param name="ssl-key-path" value=""/>
<param name="ssl-key-password" value=""/>
<!-- SSL version. If specified, must be either "SSLv3" or "TLSv1". -->
<param name="ssl-version" value=""/>
<param name="enable-ssl-verifyhost" value="false"/>
<param name="ssl-cert-path" value=""/>
<param name="enable-cacert-check" value="false"/>
<param name="ssl-cacert-file" value=""/>
</settings>
</configuration>
</include>

56
src/mod/loggers/mod_console/conf/autoload_configs/console.conf.xml Normal file
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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>

29
src/mod/loggers/mod_logfile/conf/autoload_configs/logfile.conf.xml Normal file
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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>

18
src/mod/loggers/mod_syslog/conf/autoload_configs/syslog.conf.xml Normal file
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<configuration name="syslog.conf" description="Syslog Logger">
<!-- SYSLOG -->
<!-- emerg - system is unusable -->
<!-- alert - action must be taken immediately -->
<!-- crit - critical conditions -->
<!-- err - error conditions -->
<!-- warning - warning conditions -->
<!-- notice - normal, but significant, condition -->
<!-- info - informational message -->
<!-- debug - debug-level message -->
<settings>
<param name="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>

71
src/mod/xml_int/mod_xml_cdr/conf/autoload_configs/xml_cdr.conf.xml Normal file
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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>

49
src/mod/xml_int/mod_xml_curl/conf/autoload_configs/xml_curl.conf.xml Normal file
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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>

10
src/mod/xml_int/mod_xml_rpc/conf/autoload_configs/xml_rpc.conf.xml Normal file
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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>